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  <changelog author="Benjamin A. Beasley &lt;code@musicinmybrain.net&gt; - 20240116.2-1" date="1712664000">- Update to 20240116.2 (close RHBZ#2274172)</changelog>
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  <changelog author="Benjamin A. Beasley &lt;code@musicinmybrain.net&gt; - 20240722.1-1" date="1737720000">- Update to 20240722.1 (close RHBZ#2341808)</changelog>
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  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.2.1-1" date="1508068800">- Initial build</changelog>
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  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.3.1-1" date="1673697600">- Updated to release 2.3.1</changelog>
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  <changelog author="Fabio Valentini &lt;decathorpe@gmail.com&gt; - 3.8.2-1" date="1710417600">- Update to version 3.8.2; Fixes RHBZ#2258897</changelog>
  <changelog author="S&#xE9;rgio M. Basto &lt;sergio@serjux.com&gt; - 3.8.2-2" date="1713441600">- Fix build for eln</changelog>
  <changelog author="S&#xE9;rgio M. Basto &lt;sergio@serjux.com&gt; - 3.8.2-3" date="1713441601">- jpegxl-0.10.x did not land on F40</changelog>
  <changelog author="Fabio Valentini &lt;decathorpe@gmail.com&gt; - 3.9.0-1" date="1715774400">- Update to version 3.9.0; Fixes RHBZ#2274372</changelog>
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  <changelog author="Robert-Andr&#xE9; Mauchin &lt;zebob.m@gmail.com&gt; - 3.9.0-5" date="1738238400">- Fix FTBFS with GCC15</changelog>
  <changelog author="Robert-Andr&#xE9; Mauchin &lt;zebob.m@gmail.com&gt; - 3.11.0-1" date="1738756800">- Update to 3.11</changelog>
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<package pkgid="5906b90b01e0ea776b769c4b736fff643e5d885f0076d491f193ad6d706a6697" name="asterisk" arch="src">
  <version epoch="0" ver="20.10.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="9c7e59a93dc4f59eb7e48b0735eda64f6ec6dad1d1f8fbb9fcfe165ec57a48fb" name="asterisk" arch="src">
  <version epoch="0" ver="20.11.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="8983325c59f0bda421f58aefbd550a20ffe8a5638231e31516ece45016745036" name="asterisk" arch="src">
  <version epoch="0" ver="20.11.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="9cfb9be4a5c0db2636d62ed401f8cc4cbab565af9121d124dd0765b67a994d30" name="asterisk" arch="src">
  <version epoch="0" ver="20.12.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="22d9f9a61f562dbbeeed936edea5a75a0590d4180d39736071834c1184b5b583" name="asterisk" arch="src">
  <version epoch="0" ver="20.13.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="62c93c17db2ad7582eed13b67631f120b0a4160041ca31478896528aef9f4a08" name="asterisk" arch="src">
  <version epoch="0" ver="20.14.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="5bccf6cc83aed8df1e31ec665094595d849d6b965c495e35cbd9526117b325cb" name="asterisk" arch="src">
  <version epoch="0" ver="20.14.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="4c71d92c982bdbfa780c3990e29b5cc7a26f2f8d5cc3f17a1d093ee313cb0dfb" name="asterisk" arch="src">
  <version epoch="0" ver="20.15.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
</package>
<package pkgid="2671ea8a0f203a0ac9b4cc47654f79c0db400289a552a9f89d88d11a0ca400cd" name="asterisk" arch="src">
  <version epoch="0" ver="20.15.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="4af005256bc315c6ba958a1cb0a5420660005b66f2551c652159225f2f025a1c" name="asterisk" arch="src">
  <version epoch="0" ver="20.15.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
</package>
<package pkgid="ad2d0a50b8e4bf0946ab8b25e938b53d8dc345f06a2db21f86a82afaf12fd660" name="asterisk" arch="src">
  <version epoch="0" ver="20.16.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
</package>
<package pkgid="27e7a22f72ca99ded36d7a43fc6db0df8f47f4917645c78a4de2c121e875a8e7" name="asterisk" arch="src">
  <version epoch="0" ver="20.17.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
</package>
<package pkgid="2be0f913c9219780f881d339f684a2b73baa246cf9f038301fe1ca3a1aa7d5ed" name="asterisk" arch="src">
  <version epoch="0" ver="20.18.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
</package>
<package pkgid="226d622e22c3b8509574b8c37fb150e923e6b09bc940ef2c7cf7809a4a356f9d" name="asterisk" arch="src">
  <version epoch="0" ver="20.18.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
</package>
<package pkgid="b134d284d7cdbf34c3b7c679a4231cc63f643c543b85297d5afecc2b773ce86e" name="asterisk" arch="src">
  <version epoch="0" ver="20.18.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="852aec03b58069b8b1ae448b10028114c5f9c673a1554bd2ba1184956ed14b26" name="asterisk" arch="src">
  <version epoch="0" ver="20.19.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of 
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been 
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="0f9c18c3a99cb300a996705212629a6040011fd6610b9166a316a96fe22a0bb3" name="asterisk" arch="src">
  <version epoch="0" ver="20.20.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.20.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.19.0...20.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="725dfda50da0a422433c2859a100d8081dd88e2d1abc174ca5b8505566796bfe" name="asterisk" arch="src">
  <version epoch="0" ver="20.20.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.20.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.19.0...20.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.1-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.1
-
- Change Log for Release asterisk-20.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.20.0...20.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 18
- Commit Authors: 6
- Issues Resolved: 0
- Security Advisories Resolved: 19
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-     chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-     Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-     NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-     Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-     Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-     Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-     Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-     Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-     Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-     closes during SDP processing
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-     Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-     Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-     SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-     Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-     in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-     Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count
-     Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-     LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-     Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-     Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-     ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-     ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-     and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-     Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-     2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-     ast_loggrabber executes python script in world writable directory(`/tmp`)
-     leading to potential privilege escalation And RCE
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Mike Bradeen: (2)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Mike Bradeen (2):
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                       address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h</changelog>
</package>
<package pkgid="fa1f99936705dc04d3b4d81dc9deb1f969af3065972888e5ab635edceac2c890" name="asterisk" arch="src">
  <version epoch="0" ver="20.4.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
</package>
<package pkgid="818ecfa54e5cea6b81e66774687e65116d4e78ba42e5dfef36b2b738a1a4ffdc" name="asterisk" arch="src">
  <version epoch="0" ver="20.5.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
</package>
<package pkgid="36b1f14f5c10cf146cd6527edd7ee1e76541f9167e568977bc60de5c415e8248" name="asterisk" arch="src">
  <version epoch="0" ver="20.5.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="32388fb5d9ba2d04ec00734f1f41897ed88c62501d35642112b5ca08e839f989" name="asterisk" arch="src">
  <version epoch="0" ver="20.5.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="27a37efdb2e7b195c6e981ce2aa7a29f17ffc1fa4d881b5adcccc6cade78a37d" name="asterisk" arch="src">
  <version epoch="0" ver="20.6.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="25fb2d748f9da17c12e5f76fade737274f035ca8fe2724dccd551a55e7c8d4ce" name="asterisk" arch="src">
  <version epoch="0" ver="20.7.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="cf52be96ad880a29873da3b034f3e2add7738e35de26bffb3911e7102cbb5447" name="asterisk" arch="src">
  <version epoch="0" ver="20.8.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="570a72bee46f1c91d7b0806c8dcbc8753293369e63950c4c24877a6d2eefa254" name="asterisk" arch="src">
  <version epoch="0" ver="20.8.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="ba413b0d8bcb4b23aebbe3afc4f0cac71906ac63c6f7ff3848b6161dee57fb10" name="asterisk" arch="src">
  <version epoch="0" ver="20.9.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="79ff66c3464f2f2a2607c0a3ff479b229cefce05dbbd8e498f6f85364be95dd8" name="asterisk" arch="src">
  <version epoch="0" ver="20.9.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="9fb4fab4375bf0f97fb41157df4c8a92ec85334b0606a755d7fdcd9bb9404c78" name="asterisk" arch="src">
  <version epoch="0" ver="20.9.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="cb13e50262c1878bf13693c9e7f2d0c8e9ac8ba91638afdf651232ca4b588a3c" name="asterisk" arch="src">
  <version epoch="0" ver="20.9.3" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="2f74c418d248b7a24caedcc7c26f1a3576adcb28f8fa84088ed2f02b363aebc1" name="asterisk" arch="src">
  <version epoch="0" ver="21.0.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
</package>
<package pkgid="8c6e729cfd0a35d6fbd2f2ba4ba475cdacbe85778e3a8f85b7093f214a7c52cf" name="asterisk" arch="src">
  <version epoch="0" ver="21.0.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="23d87919beaee2d76b51e3b0948668a13f62d41cde55efd068c30af19dd820d9" name="asterisk" arch="src">
  <version epoch="0" ver="21.0.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="b4f2b85483fae41c9099b9f53556a27672186e7f20efb1ba3102a1aef2d3c3ea" name="asterisk" arch="src">
  <version epoch="0" ver="21.1.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="374b9d031bdf8f812c0eebe7fb909b840e38ce151eda1b54a4b1420f762ed28e" name="asterisk" arch="src">
  <version epoch="0" ver="21.10.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
</package>
<package pkgid="d5bf6ce378d0198bfdc7adfac7be34458b374d25b73d99e3b8caa322948302bf" name="asterisk" arch="src">
  <version epoch="0" ver="21.10.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="7999db683fa5be6b07cf2fce5e97f37d639cabc3d3c2eedc8e3e1e8f07fccec9" name="asterisk" arch="src">
  <version epoch="0" ver="21.10.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
</package>
<package pkgid="cbd42a8cf91422442cf07a438044db78bfeaa8b91172197850f53a69bb6cddd4" name="asterisk" arch="src">
  <version epoch="0" ver="21.11.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
</package>
<package pkgid="345fecc6871ea391bf7a3bb7aadabe4bc56912cc26d71afc525df11bfdfe734d" name="asterisk" arch="src">
  <version epoch="0" ver="21.12.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
</package>
<package pkgid="909125d7f4459ba705477f4644e4e929ac3597e1aee0500279c2a8c5b17dad60" name="asterisk" arch="src">
  <version epoch="0" ver="21.12.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.1-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.12.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.1
-
- Change Log for Release asterisk-21.12.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.1.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.0...21.12.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-      Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-      Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-      ast_coredumper running as root sources ast_debug_tools.conf from
-      /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-      The Asterisk embedded web server's /httpstatus page echos user supplied
-      values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-      ast_coredumper runs as root, and writes gdb init file to world writeable
-      folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="bce39259d3e92a37f30e4e3fae383220d54d8b16b98bf567f41aa4b09ab52ea8" name="asterisk" arch="src">
  <version epoch="0" ver="21.12.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.1-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.12.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.1
-
- Change Log for Release asterisk-21.12.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.1.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.0...21.12.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-      Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-      Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-      ast_coredumper running as root sources ast_debug_tools.conf from
-      /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-      The Asterisk embedded web server's /httpstatus page echos user supplied
-      values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-      ast_coredumper runs as root, and writes gdb init file to world writeable
-      folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.2-1" date="1774526400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.2
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.1...21.12.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Mike Bradeen: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1833: [bug]: Address security vulnerabilities in pjproject
-
- Commits By Author:
- ------------------
-
- Mike Bradeen (1):
-  - res_pjsip: Address pjproject security vulnerabilities
-
- Commit List:
- ------------
-
- res_pjsip: Address pjproject security vulnerabilities
-
- Commit Details:
- ---------------
-
- res_pjsip: Address pjproject security vulnerabilities
-            Author: Mike Bradeen
-            Date:   2026-03-25
-
-            Address the following pjproject security vulnerabilities
-
-            [GHSA-j29p-pvh2-pvqp - Buffer overflow in ICE with long username](https://github.com/pjsip/pjproject/security/advisories/GHSA-j29p-pvh2-pvqp)
-            [GHSA-8fj4-fv9f-hjpc - Heap use-after-free in PJSIP presense subscription termination header](https://github.com/pjsip/pjproject/security/advisories/GHSA-8fj4-fv9f-hjpc)
-            [GHSA-g88q-c2hm-q7p7 - ICE session use-after-free race conditions](https://github.com/pjsip/pjproject/security/advisories/GHSA-g88q-c2hm-q7p7)
-            [GHSA-x5pq-qrp4-fmrj - Out-of-bounds read in SIP multipart parsing](https://github.com/pjsip/pjproject/security/advisories/GHSA-x5pq-qrp4-fmrj)
-
-            Resolves: #1833</changelog>
</package>
<package pkgid="ffb75ad6704369c2a0ddcfce9ab6d9bb2bd6496601204c9697d43ce94c02d79c" name="asterisk" arch="src">
  <version epoch="0" ver="21.12.3" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.1-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.12.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.1
-
- Change Log for Release asterisk-21.12.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.1.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.0...21.12.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-      Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-      Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-      ast_coredumper running as root sources ast_debug_tools.conf from
-      /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-      The Asterisk embedded web server's /httpstatus page echos user supplied
-      values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-      ast_coredumper runs as root, and writes gdb init file to world writeable
-      folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.2-1" date="1774526400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.2
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.1...21.12.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Mike Bradeen: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1833: [bug]: Address security vulnerabilities in pjproject
-
- Commits By Author:
- ------------------
-
- Mike Bradeen (1):
-  - res_pjsip: Address pjproject security vulnerabilities
-
- Commit List:
- ------------
-
- res_pjsip: Address pjproject security vulnerabilities
-
- Commit Details:
- ---------------
-
- res_pjsip: Address pjproject security vulnerabilities
-            Author: Mike Bradeen
-            Date:   2026-03-25
-
-            Address the following pjproject security vulnerabilities
-
-            [GHSA-j29p-pvh2-pvqp - Buffer overflow in ICE with long username](https://github.com/pjsip/pjproject/security/advisories/GHSA-j29p-pvh2-pvqp)
-            [GHSA-8fj4-fv9f-hjpc - Heap use-after-free in PJSIP presense subscription termination header](https://github.com/pjsip/pjproject/security/advisories/GHSA-8fj4-fv9f-hjpc)
-            [GHSA-g88q-c2hm-q7p7 - ICE session use-after-free race conditions](https://github.com/pjsip/pjproject/security/advisories/GHSA-g88q-c2hm-q7p7)
-            [GHSA-x5pq-qrp4-fmrj - Out-of-bounds read in SIP multipart parsing](https://github.com/pjsip/pjproject/security/advisories/GHSA-x5pq-qrp4-fmrj)
-
-            Resolves: #1833</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.3-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.3
-
-  Change Log for Release asterisk-21.12.3
- ========================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.3.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.2...21.12.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 21
- Commit Authors: 7
- Issues Resolved: 0
- Security Advisories Resolved: 20
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-   chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-   Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-   NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-   Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-   Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-   Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-   Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-   Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-   Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-   closes during SDP processing
-  [GHSA-h5hv-jmgj-92q2](https://github.com/asterisk/asterisk/security/advisories/GHSA-h5hv-jmgj-92q2):
-   CVE-2022-37325 fix is absent from current chan_ooh323 Q.931 party-number
-   parser
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-   Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-   Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-   SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-   Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-   in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-   Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-   LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-   Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-   Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-   ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-   ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-   and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-   Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-   2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-   ast_loggrabber executes python script in world writable directory(`/tmp`)
-   leading to potential privilege escalation And RCE
-
- User Notes:
- -----------
-
- acl: Add ACL support to http and ari
-      A new section, type=restriction has been added to http.conf to allow an
-      uri prefix based acl to be configured. See http.conf.sample for examples
-      and more information. The user section of ari.conf can now contain an acl
-      configuration to restrict users access. See ari.conf.sample for examples
-      and more information
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Joshua C. Colp: (1)
- Mike Bradeen: (4)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-h5hv-jmgj-92q2: CVE-2022-37325 fix is absent from current chan_ooh323
-                       Q.931 party-number parser
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Joshua C. Colp (1):
-  build: Fix GCC discarded-qualifiers const errors.
-
- Mike Bradeen (4):
-  ooh323c: not checking for IE minimum length
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  acl: Add ACL support to http and ari
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  ooh323c: not checking for IE minimum length
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-  acl: Add ACL support to http and ari
-  build: Fix GCC discarded-qualifiers const errors.
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- ooh323c: not checking for IE minimum length
-          Author: Mike Bradeen
-          Date:   2022-06-06
-
-          When decoding q.931 encoded calling/called number now checking for
-          length being less than minimum required.
-
-          Resolves: #GHSA-h5hv-jmgj-92q2
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h
-
- acl: Add ACL support to http and ari
-      Author: Mike Bradeen
-      Date:   2026-02-27
-
-      Add uri prefix based acl support to the built in http server. This allows
-      an acl to be added per uri prefix (ie '/metrics' or '/ws') to restrict
-      access.
-
-      Add user based acl support for ARI. This adds new acl options to the user
-      section of ari.conf to restrict access on a per user basis.
-
-      resolves: #1799
-
-      UserNote: A new section, type=restriction has been added to http.conf to
-      allow an uri prefix based acl to be configured. See http.conf.sample for
-      examples and more information. The user section of ari.conf can now
-      contain an acl configuration to restrict users access. See ari.conf.sample
-      for examples and more information
-
- build: Fix GCC discarded-qualifiers const errors.
-        Author: Joshua C. Colp
-        Date:   2026-02-12
-
-        GCC 15.2.1 pays attention to the discarding of the const qualifier when
-        strchr, strrchr, memchr, or memrchr are now used. This change fixes
-        numerous errors with this throughout the tree. The fixes can be broken
-        down into the following:
-
-        1. The return value should be considered const.
-        2. The value passed to strchr or strrchr can be cast as it is expected
-           and allowed to be modified.
-        3. The pointer passed to strchr or strrchr is not meant to be modified
-           and so the contents must be duplicated.
-        4. It was declared const and never should have been.</changelog>
</package>
<package pkgid="f4fe89027fdb2717d7398e02eef04dd083e28d2cd9fd9fbb018018bdfcd9db90" name="asterisk" arch="src">
  <version epoch="0" ver="21.2.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="c9e3cd158391cccc3a39e2d01fbc60639c5fd2bda614beeec0e37e75616a7a91" name="asterisk" arch="src">
  <version epoch="0" ver="21.3.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="dabeb77c333a4f17645da371d8b479464e308f5840ba8e4891ff1af8f51c3b1d" name="asterisk" arch="src">
  <version epoch="0" ver="21.3.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="7126025a5e6c8115bed1fca500ee89becee41f4c8879c97992a09190e3934297" name="asterisk" arch="src">
  <version epoch="0" ver="21.4.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk                             
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log             
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"                                 
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits                           
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log             
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="953601cff3ae742c6d68267010bb1a592214fd1d25166b57436149cf6dda8374" name="asterisk" arch="src">
  <version epoch="0" ver="21.4.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="2283ae60bbbb7de08c532ed22a9346cf2d28be30668330bc878f0053769baa84" name="asterisk" arch="src">
  <version epoch="0" ver="21.4.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630400">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="1e779774b0725c2dbc53f246cd2ff322cc7ca4cd7ce75f575d2ee59feba7d769" name="asterisk" arch="src">
  <version epoch="0" ver="21.4.3" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="68179c4b2bbb9ecc078862decb415de4eb76ba8364b81800b6c1de75b98923a0" name="asterisk" arch="src">
  <version epoch="0" ver="21.5.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="8418151cd40f365ffa6b55d8586e15b5862a70c873410cbbef5721175fb60fcd" name="asterisk" arch="src">
  <version epoch="0" ver="21.6.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="a06f924a0a1186447971bc2b327bfd9d3e2c7c1a58b891fa906ee5f642a5af58" name="asterisk" arch="src">
  <version epoch="0" ver="21.6.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="0a82363d229b6ace61ccdef94308447b3e3634ec4d37a2231203029235b38d79" name="asterisk" arch="src">
  <version epoch="0" ver="21.7.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="42ac2c4c879510b926e53ae405e0dfac3545583b09d35a75340e00bf54c98dff" name="asterisk" arch="src">
  <version epoch="0" ver="21.8.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="c790e40d1c53f3d08b9dc56d171559e8e508527e53808954507cf6d2fb870446" name="asterisk" arch="src">
  <version epoch="0" ver="21.9.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="b8b58a85c961e262e0041857d6f8c0d315d70058ad5953a857b8569e9fedfc85" name="asterisk" arch="src">
  <version epoch="0" ver="21.9.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="e03a58f1b504c5ee7e9f96e85e0e77369ed623b98d6745f77c66573871928fde" name="asterisk" arch="src">
  <version epoch="0" ver="22.0.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
</package>
<package pkgid="74d3cff69784b5911efef8ed51493c5d1639d9e5a0de979ac6cf83354382aaef" name="asterisk" arch="src">
  <version epoch="0" ver="22.1.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="0d473823f72f8268004202280248ac18f955c3252d8689144a01e08115b6d858" name="asterisk" arch="src">
  <version epoch="0" ver="22.1.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="6e02c8400d78262fcea441039c43e7c3baf0045973c7cfd6db729ad1063e46ff" name="asterisk" arch="src">
  <version epoch="0" ver="22.10.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.9.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.9.0
- ======================================
-
- Links:
- ------
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.2...22.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.10.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.9.0...22.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="91edf44455f9c3d2fa2b87b933b2cb44f3ee06996cb68d4767236f19a6f3bce9" name="asterisk" arch="src">
  <version epoch="0" ver="22.10.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.9.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.9.0
- ======================================
-
- Links:
- ------
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.2...22.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.10.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.9.0...22.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.10.1-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk 22.10.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.10.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.10.1
-
- Change Log for Release asterisk-22.10.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.10.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.10.0...22.10.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.10.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 6
- Issues Resolved: 0
- Security Advisories Resolved: 20
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-   chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-   Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-   NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-   Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-   Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-   Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-   Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-   Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-   Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-   closes during SDP processing
-  [GHSA-h5hv-jmgj-92q2](https://github.com/asterisk/asterisk/security/advisories/GHSA-h5hv-jmgj-92q2):
-   CVE-2022-37325 fix is absent from current chan_ooh323 Q.931 party-number
-   parser
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-   Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-   Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-   SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-   Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-   in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-   Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-   LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-   Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-   Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-   ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-   ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-   and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-   Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-   2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-   ast_loggrabber executes python script in world writable directory(`/tmp`)
-   leading to potential privilege escalation And RCE
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Mike Bradeen: (3)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-h5hv-jmgj-92q2: CVE-2022-37325 fix is absent from current chan_ooh323
-                       Q.931 party-number parser
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Mike Bradeen (3):
-  ooh323c: not checking for IE minimum length
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  ooh323c: not checking for IE minimum length
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- ooh323c: not checking for IE minimum length
-          Author: Mike Bradeen
-          Date:   2022-06-06
-
-          When decoding q.931 encoded calling/called number now checking for
-          length being less than minimum required.
-
-          Resolves: #GHSA-h5hv-jmgj-92q2
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                       address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h</changelog>
</package>
<package pkgid="e780d2868aa1745cd78e64d3f73b868c8862f8ba8dd09a73152c0ebe503e43e4" name="asterisk" arch="src">
  <version epoch="0" ver="22.2.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="13784d68856f2cb07964161c7d4c4c5f755c0d90bf8f6c6d6ea2defe15cb78f7" name="asterisk" arch="src">
  <version epoch="0" ver="22.3.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="d234679942ce0f8c3f854a6a2c9d4c6e67e0fcf995de8772b2e8af065114c618" name="asterisk" arch="src">
  <version epoch="0" ver="22.4.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="b02fadb6c70ae0da053f45ca367eb2a9e40ba466ed70efb3f8987305d7d2210e" name="asterisk" arch="src">
  <version epoch="0" ver="22.4.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="cc247f4b06be9edb1a6e6a85d553f30ef1c4d03791bc8e46590cd52d79df688d" name="asterisk" arch="src">
  <version epoch="0" ver="22.5.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
</package>
<package pkgid="6fbe744701e14eadf5a9d9e8f5d7fa279267bb71cd4634a1c152dfa718c200e2" name="asterisk" arch="src">
  <version epoch="0" ver="22.5.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="6da826044e1d3b38acc0564c7b7334de14a64d6f7167b6c649287fca769a0b15" name="asterisk" arch="src">
  <version epoch="0" ver="22.5.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
</package>
<package pkgid="3991d11a44bfdc5bb3db7463f39f2d3fc6a221137e1806b8796b4c8dd35d37b6" name="asterisk" arch="src">
  <version epoch="0" ver="22.6.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-            The new "suppress_moh_on_sendonly" endpoint option can be used
-            to prevent playing MOH back to a caller if the remote end sends
-            "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- app_mixmonitor: Add 'D' option for dual-channel audio.
-                 The MixMonitor application now has a new 'D' option which
-                 interleaves the recorded audio in the output frames. This
-                 allows for stereo recording output with one channel being
-                 the transmitted audio and the other being the received
-                 audio. The 't' and 't' options are compatible with this.
-
- manager.c: Restrict ModuleLoad to the configured modules directory.
-            The ModuleLoad AMI action now restricts modules to the
-            configured modules directory.
-
- manager: Enhance event filtering for performance
-          You can now perform more granular filtering on events in
-          manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-          = PJSIP/` This is much more efficient than `eventfilter = Event:
-          Newchannel.*Channel: PJSIP/` Full syntax guide is in
-          configs/samples/manager.conf.sample.
-
- db.c: Remove limit on family/key length
-       The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-       removed and will now accept families and keys with a total length of
-       up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-       dialplan applications, dialplan functions, manager actions and
-       `databse` CLI commands. Since the media_cache also uses the
-       `ast_db_*()` APIs, you can now store resources with URIs longer than
-       253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- manager.c: Restrict ListCategories to the configuration directory
-            The ListCategories AMI action now restricts files to the
-            configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-            Author: Ben Ford
-            Date:   2024-12-17
-
-            When using the ListCategories AMI action, it was possible to
-            traverse upwards through the directories to files outside of
-            the configured configuration directory. This action is now
-            restricted to the configured directory and an error will now be
-            returned if the specified file is outside of this limitation.
-
-            Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
</package>
<package pkgid="0c08942bda4f567fd8b92797c675d06e417d1d70dcf14e5ee91e0066c49ff1bc" name="asterisk" arch="src">
  <version epoch="0" ver="22.7.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- manager.c: Restrict ListCategories to the configuration directory
-            The ListCategories AMI action now restricts files to the
-            configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-            Author: Ben Ford
-            Date:   2024-12-17
-
-            When using the ListCategories AMI action, it was possible to
-            traverse upwards through the directories to files outside of
-            the configured configuration directory. This action is now
-            restricted to the configured directory and an error will now be
-            returned if the specified file is outside of this limitation.
-
-            Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
</package>
<package pkgid="bf7d25aae8ef1261317913737ac350399176bab14a9154e004e91be90dd2eeac" name="asterisk" arch="src">
  <version epoch="0" ver="22.8.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-                
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
</package>
<package pkgid="91dff974189c17354be58a81858ae1e10d852a1a7c6fbbfade46ea922e9212d5" name="asterisk" arch="src">
  <version epoch="0" ver="22.8.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-                
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
</package>
<package pkgid="dbc71e1a7ea5802b157f38c2a3d25dab9d03b3a8342c3977456e05bb854102db" name="asterisk" arch="src">
  <version epoch="0" ver="22.8.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="d7bcbced8696bbe3311762461035a1ded8754ac804790454df20bd94ec91c8c7" name="asterisk" arch="src">
  <version epoch="0" ver="22.9.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.9.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.9.0
- ======================================
-
- Links:
- ------
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.2...22.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="8148070ca903f30019abf8865862a9d45a10c5f585d4cff7a7cbbdd14b7ba778" name="asterisk" arch="src">
  <version epoch="0" ver="23.0.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- manager.c: Restrict ListCategories to the configuration directory
-            The ListCategories AMI action now restricts files to the
-            configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-            Author: Ben Ford
-            Date:   2024-12-17
-
-            When using the ListCategories AMI action, it was possible to
-            traverse upwards through the directories to files outside of
-            the configured configuration directory. This action is now
-            restricted to the configured directory and an error will now be
-            returned if the specified file is outside of this limitation.
-
-            Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
</package>
<package pkgid="9ac66d826aefd6d3777d373ece3b604529e23354598ea8c8c11fa63edaa201ff" name="asterisk" arch="src">
  <version epoch="0" ver="23.1.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
</package>
<package pkgid="1e1000c22a9664bcbf3339e83921546c3fd4a89e76d3b227a6d05b970205bd9f" name="asterisk" arch="src">
  <version epoch="0" ver="23.2.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
</package>
<package pkgid="92bd0f44eeb7e19ff77ff1e3ff45b59c0428d9b3196eec2c12d12e104ca8be28" name="asterisk" arch="src">
  <version epoch="0" ver="23.2.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
</package>
<package pkgid="32eb52b17ac5f620cc39aa68d6f9164974a7139feb34adb73a8a2ebb0933e237" name="asterisk" arch="src">
  <version epoch="0" ver="23.2.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="f59a1763bbc19c1738475ae84546c834556ff2f274a835afd199355fc95f58df" name="asterisk" arch="src">
  <version epoch="0" ver="23.3.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.3.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.2...23.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="e120a5786b6525c166b7ba0d1e90cea6a51f01ac5de3fd5131a478c0705be8b9" name="asterisk" arch="src">
  <version epoch="0" ver="23.4.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.3.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.2...23.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.4.0-1" date="1775736001">- The Asterisk Development Team would like to announce the release of
- asterisk-23.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.3.0...23.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="d92b739eecd8d48d28306239dee57283731ce2b434cda733b88a3900f2919c2e" name="asterisk" arch="src">
  <version epoch="0" ver="23.4.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.3.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.2...23.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.4.0-1" date="1775736001">- The Asterisk Development Team would like to announce the release of
- asterisk-23.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.3.0...23.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.4.1-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.4.1
-
- Change Log for Release asterisk-23.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.4.0...23.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 6
- Issues Resolved: 0
- Security Advisories Resolved: 20
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-   chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-   Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-   NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-   Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-   Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-   Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-   Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-   Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-   Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-   closes during SDP processing
-  [GHSA-h5hv-jmgj-92q2](https://github.com/asterisk/asterisk/security/advisories/GHSA-h5hv-jmgj-92q2):
-   CVE-2022-37325 fix is absent from current chan_ooh323 Q.931 party-number
-   parser
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-   Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-   Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-   SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-   Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-   in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-   Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-   LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-   Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-   Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-   ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-   ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-   and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-   Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-   2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-   ast_loggrabber executes python script in world writable directory(`/tmp`)
-   leading to potential privilege escalation And RCE
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Mike Bradeen: (3)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-h5hv-jmgj-92q2: CVE-2022-37325 fix is absent from current chan_ooh323
-                       Q.931 party-number parser
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Mike Bradeen (3):
-  ooh323c: not checking for IE minimum length
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- #### ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  ooh323c: not checking for IE minimum length
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- ooh323c: not checking for IE minimum length
-          Author: Mike Bradeen
-          Date:   2022-06-06
-
-          When decoding q.931 encoded calling/called number now checking for
-          length being less than minimum required.
-
-          Resolves: #GHSA-h5hv-jmgj-92q2
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                       address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h</changelog>
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- Add opional check
- Fix license</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 23.2.14-3.giteb4a667.1" date="1705665600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_40_Mass_Rebuild</changelog>
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  <changelog author="Tom Rix &lt;trix@redhat.com&gt; - 23.11.04-0.gitd6860c4" date="1710072000">- Update for pytorch 2.3</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 23.11.04-0.gitd6860c4.1" date="1721217600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_41_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 24.09.26-1.git1e83a2f.1" date="1737028800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_42_Mass_Rebuild</changelog>
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  See https://pagure.io/fesco/issue/2583.</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.11.1-23" date="1626868800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_35_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.11.1-24" date="1642680000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_36_Mass_Rebuild</changelog>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 2.11.1-25" date="1654084800">- Perl 5.36 rebuild</changelog>
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- Move configure to build section because of linker package notes (bug #2045296)
- Respect distribution CFLAGS
- Set Perl libdir with a configure option</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.11.1-27" date="1658318400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_37_Mass_Rebuild</changelog>
  <changelog author="Davide Cavalca &lt;dcavalca@fedoraproject.org&gt; - 2.11.1-28" date="1661515200">- Adjust libusb-devel BuildRequires to fix build on EPEL</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.3.0-3" date="1706097600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_40_Mass_Rebuild</changelog>
  <changelog author="Fabio Valentini &lt;decathorpe@gmail.com&gt; - 1.4.0-1" date="1709208000">- Update to version 1.4.0; Fixes RHBZ#2264274</changelog>
  <changelog author="Fabio Valentini &lt;decathorpe@gmail.com&gt; - 1.4.3-1" date="1718366400">- Update to version 1.4.3; Fixes RHBZ#2269666</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.4.3-2" date="1721217600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_41_Mass_Rebuild</changelog>
  <changelog author="Peter Robinson &lt;pbrobinson@gmail.com&gt; - 1.5.0-1" date="1729339200">- Update to 1.5.0</changelog>
  <changelog author="Fabio Valentini &lt;decathorpe@gmail.com&gt; - 1.5.0-2" date="1729512000">- Revert unwanted changes to .gitignore</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.5.0-3" date="1737028800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_42_Mass_Rebuild</changelog>
  <changelog author="Fabio Valentini &lt;decathorpe@gmail.com&gt; - 1.5.1-1" date="1737892800">- Update to version 1.5.1; Fixes RHBZ#2338924</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.2.7-4" date="1674129600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_38_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.2.7-5" date="1689768000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_39_Mass_Rebuild</changelog>
  <changelog author="Greg Hellings &lt;greg.hellings@gmail.com&gt; - 2.2.7-6" date="1701172800">- Use upstream manpage install command
- Rename man pages whose names are bad</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.2.7-7" date="1705665600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_40_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.2.7-8" date="1706097600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_40_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.2.7-9" date="1721217600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_41_Mass_Rebuild</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.1.0-0.20.20070619svnrev12" date="1689768000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_39_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.1.0-0.21.20070619svnrev12" date="1705665600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_40_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.1.0-0.22.20070619svnrev12" date="1706097600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_40_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.1.0-0.23.20070619svnrev12" date="1721217600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_41_Mass_Rebuild</changelog>
  <changelog author="Remi Collet &lt;remi@remirepo.net&gt; - 0.5.0-1" date="1729080000">- update to 0.5.0</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.00.38-4" date="1501675200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Binutils_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.00.38-5" date="1518004800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.00.38-6" date="1531483200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.00.38-7" date="1548936000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
  <changelog author="Igor Gnatenko &lt;ignatenkobrain@fedoraproject.org&gt; - 1.00.38-8" date="1550404800">- Rebuild for readline 8.0</changelog>
  <changelog author="Dmitry Butskoy &lt;Dmitry@Butskoy.name&gt; - 1.1.11-1" date="1562673600">- Upgrade to 1.1.11 (#1728191)</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.00.38-5" date="1518004800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.00.38-6" date="1531483200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.00.38-7" date="1548936000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
  <changelog author="Igor Gnatenko &lt;ignatenkobrain@fedoraproject.org&gt; - 1.00.38-8" date="1550404800">- Rebuild for readline 8.0</changelog>
  <changelog author="Dmitry Butskoy &lt;Dmitry@Butskoy.name&gt; - 1.1.11-1" date="1562673600">- Upgrade to 1.1.11 (#1728191)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.12-1" date="1565438400">- Upgrade to 1.1.12</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.00.38-6" date="1531483200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.00.38-7" date="1548936000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
  <changelog author="Igor Gnatenko &lt;ignatenkobrain@fedoraproject.org&gt; - 1.00.38-8" date="1550404800">- Rebuild for readline 8.0</changelog>
  <changelog author="Dmitry Butskoy &lt;Dmitry@Butskoy.name&gt; - 1.1.11-1" date="1562673600">- Upgrade to 1.1.11 (#1728191)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.12-1" date="1565438400">- Upgrade to 1.1.12</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.14-1" date="1567252800">- Upgrade to 1.1.14</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.00.38-5" date="1518004800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.00.38-6" date="1531483200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.00.38-7" date="1548936000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
  <changelog author="Igor Gnatenko &lt;ignatenkobrain@fedoraproject.org&gt; - 1.00.38-8" date="1550404800">- Rebuild for readline 8.0</changelog>
  <changelog author="Dmitry Butskoy &lt;Dmitry@Butskoy.name&gt; - 1.1.11-1" date="1562673600">- Upgrade to 1.1.11 (#1728191)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.12-1" date="1565438400">- Upgrade to 1.1.12</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.15-1" date="1567425600">- Upgrade to 1.1.15</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.00.38-5" date="1518004800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.00.38-6" date="1531483200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.00.38-7" date="1548936000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
  <changelog author="Igor Gnatenko &lt;ignatenkobrain@fedoraproject.org&gt; - 1.00.38-8" date="1550404800">- Rebuild for readline 8.0</changelog>
  <changelog author="Dmitry Butskoy &lt;Dmitry@Butskoy.name&gt; - 1.1.11-1" date="1562673600">- Upgrade to 1.1.11 (#1728191)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.12-1" date="1565438400">- Upgrade to 1.1.12</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.15-1" date="1567425600">- Upgrade to 1.1.15</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.16-1" date="1569240000">- Upgrade to 1.1.16</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.00.38-5" date="1518004800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.00.38-6" date="1531483200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.00.38-7" date="1548936000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
  <changelog author="Igor Gnatenko &lt;ignatenkobrain@fedoraproject.org&gt; - 1.00.38-8" date="1550404800">- Rebuild for readline 8.0</changelog>
  <changelog author="Dmitry Butskoy &lt;Dmitry@Butskoy.name&gt; - 1.1.11-1" date="1562673600">- Upgrade to 1.1.11 (#1728191)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.12-1" date="1565438400">- Upgrade to 1.1.12</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.15-1" date="1567425600">- Upgrade to 1.1.15</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.16-1" date="1569240000">- Upgrade to 1.1.16</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.17-1" date="1570536000">- Upgrade to 1.1.17</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.00.38-5" date="1518004800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.00.38-6" date="1531483200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.00.38-7" date="1548936000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
  <changelog author="Igor Gnatenko &lt;ignatenkobrain@fedoraproject.org&gt; - 1.00.38-8" date="1550404800">- Rebuild for readline 8.0</changelog>
  <changelog author="Dmitry Butskoy &lt;Dmitry@Butskoy.name&gt; - 1.1.11-1" date="1562673600">- Upgrade to 1.1.11 (#1728191)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.12-1" date="1565438400">- Upgrade to 1.1.12</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.15-1" date="1567425600">- Upgrade to 1.1.15</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.16-1" date="1569240000">- Upgrade to 1.1.16</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.17-1" date="1570536000">- Upgrade to 1.1.17</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.19-1" date="1570968000">- Upgrade to 1.1.19</changelog>
</package>
<package pkgid="ed8eef2a41f10185e939ef0cb5fa49790c64dcce4fdaeb2a6661b4a2658ff00d" name="freetds" arch="src">
  <version epoch="0" ver="1.1.20" rel="1.el10"/>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.00.38-6" date="1531483200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.00.38-7" date="1548936000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
  <changelog author="Igor Gnatenko &lt;ignatenkobrain@fedoraproject.org&gt; - 1.00.38-8" date="1550404800">- Rebuild for readline 8.0</changelog>
  <changelog author="Dmitry Butskoy &lt;Dmitry@Butskoy.name&gt; - 1.1.11-1" date="1562673600">- Upgrade to 1.1.11 (#1728191)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.12-1" date="1565438400">- Upgrade to 1.1.12</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.15-1" date="1567425600">- Upgrade to 1.1.15</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.16-1" date="1569240000">- Upgrade to 1.1.16</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.17-1" date="1570536000">- Upgrade to 1.1.17</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.19-1" date="1570968000">- Upgrade to 1.1.19</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.20-1" date="1571313600">- Upgrade to 1.1.20</changelog>
</package>
<package pkgid="7ef17855fd1891388f72f9c9f7f5f103f7cbaab92619fcde14cbe20e3fad474c" name="freetds" arch="src">
  <version epoch="0" ver="1.1.21" rel="1.el10"/>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.00.38-7" date="1548936000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
  <changelog author="Igor Gnatenko &lt;ignatenkobrain@fedoraproject.org&gt; - 1.00.38-8" date="1550404800">- Rebuild for readline 8.0</changelog>
  <changelog author="Dmitry Butskoy &lt;Dmitry@Butskoy.name&gt; - 1.1.11-1" date="1562673600">- Upgrade to 1.1.11 (#1728191)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.12-1" date="1565438400">- Upgrade to 1.1.12</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.15-1" date="1567425600">- Upgrade to 1.1.15</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.16-1" date="1569240000">- Upgrade to 1.1.16</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.17-1" date="1570536000">- Upgrade to 1.1.17</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.19-1" date="1570968000">- Upgrade to 1.1.19</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.20-1" date="1571313600">- Upgrade to 1.1.20</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.21-1" date="1573646400">- Upgrade to 1.1.21</changelog>
</package>
<package pkgid="c47390ea7820aa664151afb12072b262221d4f2e2081e19a21f19dbd4ec73063" name="freetds" arch="src">
  <version epoch="0" ver="1.1.23" rel="1.el10"/>
  <changelog author="Igor Gnatenko &lt;ignatenkobrain@fedoraproject.org&gt; - 1.00.38-8" date="1550404800">- Rebuild for readline 8.0</changelog>
  <changelog author="Dmitry Butskoy &lt;Dmitry@Butskoy.name&gt; - 1.1.11-1" date="1562673600">- Upgrade to 1.1.11 (#1728191)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.12-1" date="1565438400">- Upgrade to 1.1.12</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.15-1" date="1567425600">- Upgrade to 1.1.15</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.16-1" date="1569240000">- Upgrade to 1.1.16</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.17-1" date="1570536000">- Upgrade to 1.1.17</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.19-1" date="1570968000">- Upgrade to 1.1.19</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.20-1" date="1571313600">- Upgrade to 1.1.20</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.21-1" date="1573646400">- Upgrade to 1.1.21</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.23-1" date="1575115200">- Upgrade to 1.1.23</changelog>
</package>
<package pkgid="a7a4e5cc836fbd810273cf97643f4816e72acd3aab1becdf95760acd1a073991" name="freetds" arch="src">
  <version epoch="0" ver="1.1.24" rel="1.el10"/>
  <changelog author="Igor Gnatenko &lt;ignatenkobrain@fedoraproject.org&gt; - 1.00.38-8" date="1550404800">- Rebuild for readline 8.0</changelog>
  <changelog author="Dmitry Butskoy &lt;Dmitry@Butskoy.name&gt; - 1.1.11-1" date="1562673600">- Upgrade to 1.1.11 (#1728191)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.12-1" date="1565438400">- Upgrade to 1.1.12</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.15-1" date="1567425600">- Upgrade to 1.1.15</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.16-1" date="1569240000">- Upgrade to 1.1.16</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.17-1" date="1570536000">- Upgrade to 1.1.17</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.19-1" date="1570968000">- Upgrade to 1.1.19</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.20-1" date="1571313600">- Upgrade to 1.1.20</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.21-1" date="1573646400">- Upgrade to 1.1.21</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.24-1" date="1575979200">- Upgrade to 1.1.24</changelog>
</package>
<package pkgid="20329803a1b3048b1b904711451df0c7983413d4e844eb7fd73a8dd23b1e9900" name="freetds" arch="src">
  <version epoch="0" ver="1.1.26" rel="1.el10"/>
  <changelog author="Dmitry Butskoy &lt;Dmitry@Butskoy.name&gt; - 1.1.11-1" date="1562673600">- Upgrade to 1.1.11 (#1728191)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.12-1" date="1565438400">- Upgrade to 1.1.12</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.15-1" date="1567425600">- Upgrade to 1.1.15</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.16-1" date="1569240000">- Upgrade to 1.1.16</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.17-1" date="1570536000">- Upgrade to 1.1.17</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.19-1" date="1570968000">- Upgrade to 1.1.19</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.20-1" date="1571313600">- Upgrade to 1.1.20</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.21-1" date="1573646400">- Upgrade to 1.1.21</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.24-1" date="1575979200">- Upgrade to 1.1.24</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.26-1" date="1585569600">- Upgrade to 1.1.26</changelog>
</package>
<package pkgid="9309311b08924e67c0c41295f96225b46a19bb3e86c97c9222b47fdc08d223b0" name="freetds" arch="src">
  <version epoch="0" ver="1.1.29" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.12-1" date="1565438400">- Upgrade to 1.1.12</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.15-1" date="1567425600">- Upgrade to 1.1.15</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.16-1" date="1569240000">- Upgrade to 1.1.16</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.17-1" date="1570536000">- Upgrade to 1.1.17</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.19-1" date="1570968000">- Upgrade to 1.1.19</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.20-1" date="1571313600">- Upgrade to 1.1.20</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.21-1" date="1573646400">- Upgrade to 1.1.21</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.24-1" date="1575979200">- Upgrade to 1.1.24</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.26-1" date="1585569600">- Upgrade to 1.1.26</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.29-1" date="1586433600">- Upgrade to 1.1.29</changelog>
</package>
<package pkgid="fb2fdcf0146650c30da15083478c6551833b958b74bdba4930c1a74426288f47" name="freetds" arch="src">
  <version epoch="0" ver="1.1.30" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.15-1" date="1567425600">- Upgrade to 1.1.15</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.16-1" date="1569240000">- Upgrade to 1.1.16</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.17-1" date="1570536000">- Upgrade to 1.1.17</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.19-1" date="1570968000">- Upgrade to 1.1.19</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.20-1" date="1571313600">- Upgrade to 1.1.20</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.21-1" date="1573646400">- Upgrade to 1.1.21</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.24-1" date="1575979200">- Upgrade to 1.1.24</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.26-1" date="1585569600">- Upgrade to 1.1.26</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.29-1" date="1586433600">- Upgrade to 1.1.29</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.30-1" date="1586520000">- Upgrade to 1.1.30</changelog>
</package>
<package pkgid="2713981a3aedee40cdfb04de9c329ee0bfd2ff56c372028f9b9602dc797f422c" name="freetds" arch="src">
  <version epoch="0" ver="1.1.32" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.16-1" date="1569240000">- Upgrade to 1.1.16</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.17-1" date="1570536000">- Upgrade to 1.1.17</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.19-1" date="1570968000">- Upgrade to 1.1.19</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.20-1" date="1571313600">- Upgrade to 1.1.20</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.21-1" date="1573646400">- Upgrade to 1.1.21</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.24-1" date="1575979200">- Upgrade to 1.1.24</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.26-1" date="1585569600">- Upgrade to 1.1.26</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.29-1" date="1586433600">- Upgrade to 1.1.29</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.30-1" date="1586520000">- Upgrade to 1.1.30</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.32-1" date="1586606400">- Upgrade to 1.1.32</changelog>
</package>
<package pkgid="159da76aa0969f4c32a3709be1733d739b0553c91b926bef77b4df166f38489f" name="freetds" arch="src">
  <version epoch="0" ver="1.1.33" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.17-1" date="1570536000">- Upgrade to 1.1.17</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.19-1" date="1570968000">- Upgrade to 1.1.19</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.20-1" date="1571313600">- Upgrade to 1.1.20</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.21-1" date="1573646400">- Upgrade to 1.1.21</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.24-1" date="1575979200">- Upgrade to 1.1.24</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.26-1" date="1585569600">- Upgrade to 1.1.26</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.29-1" date="1586433600">- Upgrade to 1.1.29</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.30-1" date="1586520000">- Upgrade to 1.1.30</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.32-1" date="1586606400">- Upgrade to 1.1.32</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.33-1" date="1587124800">- Upgrade to 1.1.33</changelog>
</package>
<package pkgid="ef163e575b59992c3176009ec9e254aa0130effd860e89c0d33a25717755c7c6" name="freetds" arch="src">
  <version epoch="0" ver="1.1.36" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.19-1" date="1570968000">- Upgrade to 1.1.19</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.20-1" date="1571313600">- Upgrade to 1.1.20</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.21-1" date="1573646400">- Upgrade to 1.1.21</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.24-1" date="1575979200">- Upgrade to 1.1.24</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.26-1" date="1585569600">- Upgrade to 1.1.26</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.29-1" date="1586433600">- Upgrade to 1.1.29</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.30-1" date="1586520000">- Upgrade to 1.1.30</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.32-1" date="1586606400">- Upgrade to 1.1.32</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.33-1" date="1587124800">- Upgrade to 1.1.33</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.36-1" date="1588593600">- Upgrade to 1.1.36</changelog>
</package>
<package pkgid="72031dbda0601c680308c1502e168006a8d2e5963797985fbd65451720258a68" name="freetds" arch="src">
  <version epoch="0" ver="1.1.39" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.20-1" date="1571313600">- Upgrade to 1.1.20</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.21-1" date="1573646400">- Upgrade to 1.1.21</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.24-1" date="1575979200">- Upgrade to 1.1.24</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.26-1" date="1585569600">- Upgrade to 1.1.26</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.29-1" date="1586433600">- Upgrade to 1.1.29</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.30-1" date="1586520000">- Upgrade to 1.1.30</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.32-1" date="1586606400">- Upgrade to 1.1.32</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.33-1" date="1587124800">- Upgrade to 1.1.33</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.36-1" date="1588593600">- Upgrade to 1.1.36</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.39-1" date="1589716800">- Upgrade to 1.1.39</changelog>
</package>
<package pkgid="04fc0613393dcaec8200952a6edd6c57314ed12033c43b6942f8eef9068df8e6" name="freetds" arch="src">
  <version epoch="0" ver="1.1.40" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.21-1" date="1573646400">- Upgrade to 1.1.21</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.24-1" date="1575979200">- Upgrade to 1.1.24</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.26-1" date="1585569600">- Upgrade to 1.1.26</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.29-1" date="1586433600">- Upgrade to 1.1.29</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.30-1" date="1586520000">- Upgrade to 1.1.30</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.32-1" date="1586606400">- Upgrade to 1.1.32</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.33-1" date="1587124800">- Upgrade to 1.1.33</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.36-1" date="1588593600">- Upgrade to 1.1.36</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.39-1" date="1589716800">- Upgrade to 1.1.39</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.40-1" date="1590148800">- Upgrade to 1.1.40</changelog>
</package>
<package pkgid="2a894aeb335904ae445ae2b931436678c4e11e14bc5b1869a1bb268e93f1be13" name="freetds" arch="src">
  <version epoch="0" ver="1.1.41" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.24-1" date="1575979200">- Upgrade to 1.1.24</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.26-1" date="1585569600">- Upgrade to 1.1.26</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.29-1" date="1586433600">- Upgrade to 1.1.29</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.30-1" date="1586520000">- Upgrade to 1.1.30</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.32-1" date="1586606400">- Upgrade to 1.1.32</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.33-1" date="1587124800">- Upgrade to 1.1.33</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.36-1" date="1588593600">- Upgrade to 1.1.36</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.39-1" date="1589716800">- Upgrade to 1.1.39</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.40-1" date="1590148800">- Upgrade to 1.1.40</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.41-1" date="1592049600">- Upgrade to 1.1.41</changelog>
</package>
<package pkgid="09fe977c3bf693264c6627e6e68ed02582eba2897f275188e31b98d7febcd6ae" name="freetds" arch="src">
  <version epoch="0" ver="1.1.42" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.26-1" date="1585569600">- Upgrade to 1.1.26</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.29-1" date="1586433600">- Upgrade to 1.1.29</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.30-1" date="1586520000">- Upgrade to 1.1.30</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.32-1" date="1586606400">- Upgrade to 1.1.32</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.33-1" date="1587124800">- Upgrade to 1.1.33</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.36-1" date="1588593600">- Upgrade to 1.1.36</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.39-1" date="1589716800">- Upgrade to 1.1.39</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.40-1" date="1590148800">- Upgrade to 1.1.40</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.41-1" date="1592049600">- Upgrade to 1.1.41</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.42-1" date="1592136000">- Upgrade to 1.1.42</changelog>
</package>
<package pkgid="3f5501545cdb5203e8040cba370946a275180843c3b7688aaa008fc9239dbc60" name="freetds" arch="src">
  <version epoch="0" ver="1.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.29-1" date="1586433600">- Upgrade to 1.1.29</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.30-1" date="1586520000">- Upgrade to 1.1.30</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.32-1" date="1586606400">- Upgrade to 1.1.32</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.33-1" date="1587124800">- Upgrade to 1.1.33</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.36-1" date="1588593600">- Upgrade to 1.1.36</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.39-1" date="1589716800">- Upgrade to 1.1.39</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.40-1" date="1590148800">- Upgrade to 1.1.40</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.41-1" date="1592049600">- Upgrade to 1.1.41</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.42-1" date="1592136000">- Upgrade to 1.1.42</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2-1" date="1593777600">- Upgrade to 1.2</changelog>
</package>
<package pkgid="ec0efed81ed9a6d22cc095590704869d5c4d16095632862fb56d15a4b5eb9c1b" name="freetds" arch="src">
  <version epoch="0" ver="1.2.10" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.41-1" date="1592049600">- Upgrade to 1.1.41</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.42-1" date="1592136000">- Upgrade to 1.1.42</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2-1" date="1593777600">- Upgrade to 1.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.2-1" date="1594123200">- Upgrade to 1.2.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.3-1" date="1594296000">- Upgrade to 1.2.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.4-1" date="1599220800">- Upgrade to 1.2.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.5-1" date="1600430400">- Upgrade to 1.2.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.6-1" date="1603972800">- Upgrade to 1.2.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.9-1" date="1604059200">- Upgrade to 1.2.9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.10-1" date="1604145600">- Upgrade to 1.2.10</changelog>
</package>
<package pkgid="b9082128cf7abc95a10bf1e780f4883b3709726f7806f006fb1f70850f33d05f" name="freetds" arch="src">
  <version epoch="0" ver="1.2.11" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.42-1" date="1592136000">- Upgrade to 1.1.42</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2-1" date="1593777600">- Upgrade to 1.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.2-1" date="1594123200">- Upgrade to 1.2.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.3-1" date="1594296000">- Upgrade to 1.2.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.4-1" date="1599220800">- Upgrade to 1.2.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.5-1" date="1600430400">- Upgrade to 1.2.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.6-1" date="1603972800">- Upgrade to 1.2.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.9-1" date="1604059200">- Upgrade to 1.2.9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.10-1" date="1604145600">- Upgrade to 1.2.10</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.11-1" date="1605182400">- Upgrade to 1.2.11</changelog>
</package>
<package pkgid="61a6013d4b529b623f48c0de7852e3c742114e71c0a34066b93a80ad756a4e4f" name="freetds" arch="src">
  <version epoch="0" ver="1.2.12" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2-1" date="1593777600">- Upgrade to 1.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.2-1" date="1594123200">- Upgrade to 1.2.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.3-1" date="1594296000">- Upgrade to 1.2.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.4-1" date="1599220800">- Upgrade to 1.2.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.5-1" date="1600430400">- Upgrade to 1.2.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.6-1" date="1603972800">- Upgrade to 1.2.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.9-1" date="1604059200">- Upgrade to 1.2.9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.10-1" date="1604145600">- Upgrade to 1.2.10</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.11-1" date="1605182400">- Upgrade to 1.2.11</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.12-1" date="1605960000">- Upgrade to 1.2.12</changelog>
</package>
<package pkgid="34eed24aae1928f8b1cd51298ee029b8ac2f85e502d3fc0b0381a7c26e24b411" name="freetds" arch="src">
  <version epoch="0" ver="1.2.13" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.2-1" date="1594123200">- Upgrade to 1.2.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.3-1" date="1594296000">- Upgrade to 1.2.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.4-1" date="1599220800">- Upgrade to 1.2.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.5-1" date="1600430400">- Upgrade to 1.2.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.6-1" date="1603972800">- Upgrade to 1.2.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.9-1" date="1604059200">- Upgrade to 1.2.9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.10-1" date="1604145600">- Upgrade to 1.2.10</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.11-1" date="1605182400">- Upgrade to 1.2.11</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.12-1" date="1605960000">- Upgrade to 1.2.12</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.13-1" date="1606651200">- Upgrade to 1.2.13</changelog>
</package>
<package pkgid="82259335f0b6794c1017546c497b47401353253840748167d0d9444adda08305" name="freetds" arch="src">
  <version epoch="0" ver="1.2.17" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.3-1" date="1594296000">- Upgrade to 1.2.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.4-1" date="1599220800">- Upgrade to 1.2.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.5-1" date="1600430400">- Upgrade to 1.2.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.6-1" date="1603972800">- Upgrade to 1.2.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.9-1" date="1604059200">- Upgrade to 1.2.9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.10-1" date="1604145600">- Upgrade to 1.2.10</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.11-1" date="1605182400">- Upgrade to 1.2.11</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.12-1" date="1605960000">- Upgrade to 1.2.12</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.13-1" date="1606651200">- Upgrade to 1.2.13</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.17-1" date="1607515200">- Upgrade to 1.2.17</changelog>
</package>
<package pkgid="19ac7bfd199a901d9b4bd4bccd619133f38b540914a16796db8f188ed91b1590" name="freetds" arch="src">
  <version epoch="0" ver="1.2.18" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.4-1" date="1599220800">- Upgrade to 1.2.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.5-1" date="1600430400">- Upgrade to 1.2.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.6-1" date="1603972800">- Upgrade to 1.2.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.9-1" date="1604059200">- Upgrade to 1.2.9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.10-1" date="1604145600">- Upgrade to 1.2.10</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.11-1" date="1605182400">- Upgrade to 1.2.11</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.12-1" date="1605960000">- Upgrade to 1.2.12</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.13-1" date="1606651200">- Upgrade to 1.2.13</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.17-1" date="1607515200">- Upgrade to 1.2.17</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.18-1" date="1608292800">- Upgrade to 1.2.18</changelog>
</package>
<package pkgid="ec80b5f46b2101cafe1fb7f19b8fcbaef5f039cf3431c5abf93fd071a478783a" name="freetds" arch="src">
  <version epoch="0" ver="1.2.19" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.5-1" date="1600430400">- Upgrade to 1.2.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.6-1" date="1603972800">- Upgrade to 1.2.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.9-1" date="1604059200">- Upgrade to 1.2.9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.10-1" date="1604145600">- Upgrade to 1.2.10</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.11-1" date="1605182400">- Upgrade to 1.2.11</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.12-1" date="1605960000">- Upgrade to 1.2.12</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.13-1" date="1606651200">- Upgrade to 1.2.13</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.17-1" date="1607515200">- Upgrade to 1.2.17</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.18-1" date="1608292800">- Upgrade to 1.2.18</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.19-1" date="1618056000">- Upgrade to 1.2.19</changelog>
</package>
<package pkgid="14695a581cacb8a0abe82725f24bbe7aa9addc2a18e59b2264314b5f45eaa8ac" name="freetds" arch="src">
  <version epoch="0" ver="1.2.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.30-1" date="1586520000">- Upgrade to 1.1.30</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.32-1" date="1586606400">- Upgrade to 1.1.32</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.33-1" date="1587124800">- Upgrade to 1.1.33</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.36-1" date="1588593600">- Upgrade to 1.1.36</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.39-1" date="1589716800">- Upgrade to 1.1.39</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.40-1" date="1590148800">- Upgrade to 1.1.40</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.41-1" date="1592049600">- Upgrade to 1.1.41</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.42-1" date="1592136000">- Upgrade to 1.1.42</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2-1" date="1593777600">- Upgrade to 1.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.2-1" date="1594123200">- Upgrade to 1.2.2</changelog>
</package>
<package pkgid="08fb77c488445ed5ac69c119e45a85f5aac5b2a689c514ab7ab3a0af0bc59c49" name="freetds" arch="src">
  <version epoch="0" ver="1.2.20" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.6-1" date="1603972800">- Upgrade to 1.2.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.9-1" date="1604059200">- Upgrade to 1.2.9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.10-1" date="1604145600">- Upgrade to 1.2.10</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.11-1" date="1605182400">- Upgrade to 1.2.11</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.12-1" date="1605960000">- Upgrade to 1.2.12</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.13-1" date="1606651200">- Upgrade to 1.2.13</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.17-1" date="1607515200">- Upgrade to 1.2.17</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.18-1" date="1608292800">- Upgrade to 1.2.18</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.19-1" date="1618056000">- Upgrade to 1.2.19</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.20-1" date="1618315200">- Upgrade to 1.2.20</changelog>
</package>
<package pkgid="9e85f1fb9530050fceec86c256e4ff9fb605f5fa235cbe565b9c0e82f067f9da" name="freetds" arch="src">
  <version epoch="0" ver="1.2.21" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.9-1" date="1604059200">- Upgrade to 1.2.9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.10-1" date="1604145600">- Upgrade to 1.2.10</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.11-1" date="1605182400">- Upgrade to 1.2.11</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.12-1" date="1605960000">- Upgrade to 1.2.12</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.13-1" date="1606651200">- Upgrade to 1.2.13</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.17-1" date="1607515200">- Upgrade to 1.2.17</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.18-1" date="1608292800">- Upgrade to 1.2.18</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.19-1" date="1618056000">- Upgrade to 1.2.19</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.20-1" date="1618315200">- Upgrade to 1.2.20</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.21-1" date="1619265600">- Upgrade to 1.2.21</changelog>
</package>
<package pkgid="932f490489f6e1f2496dd38b75568679e7d1c90541bdd5c0ab9af4f9d9e57afb" name="freetds" arch="src">
  <version epoch="0" ver="1.2.3" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.32-1" date="1586606400">- Upgrade to 1.1.32</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.33-1" date="1587124800">- Upgrade to 1.1.33</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.36-1" date="1588593600">- Upgrade to 1.1.36</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.39-1" date="1589716800">- Upgrade to 1.1.39</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.40-1" date="1590148800">- Upgrade to 1.1.40</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.41-1" date="1592049600">- Upgrade to 1.1.41</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.42-1" date="1592136000">- Upgrade to 1.1.42</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2-1" date="1593777600">- Upgrade to 1.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.2-1" date="1594123200">- Upgrade to 1.2.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.3-1" date="1594296000">- Upgrade to 1.2.3</changelog>
</package>
<package pkgid="ebee10889037edf9122bbb973d6211c792b2f7d20279f2594fc6e6f46505a87e" name="freetds" arch="src">
  <version epoch="0" ver="1.2.4" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.33-1" date="1587124800">- Upgrade to 1.1.33</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.36-1" date="1588593600">- Upgrade to 1.1.36</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.39-1" date="1589716800">- Upgrade to 1.1.39</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.40-1" date="1590148800">- Upgrade to 1.1.40</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.41-1" date="1592049600">- Upgrade to 1.1.41</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.42-1" date="1592136000">- Upgrade to 1.1.42</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2-1" date="1593777600">- Upgrade to 1.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.2-1" date="1594123200">- Upgrade to 1.2.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.3-1" date="1594296000">- Upgrade to 1.2.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.4-1" date="1599220800">- Upgrade to 1.2.4</changelog>
</package>
<package pkgid="90efb5f3a4d0d4c11b7a38a15e026fa5571d9c51383fdc9f65cb22556e86e6ba" name="freetds" arch="src">
  <version epoch="0" ver="1.2.5" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.36-1" date="1588593600">- Upgrade to 1.1.36</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.39-1" date="1589716800">- Upgrade to 1.1.39</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.40-1" date="1590148800">- Upgrade to 1.1.40</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.41-1" date="1592049600">- Upgrade to 1.1.41</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.42-1" date="1592136000">- Upgrade to 1.1.42</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2-1" date="1593777600">- Upgrade to 1.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.2-1" date="1594123200">- Upgrade to 1.2.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.3-1" date="1594296000">- Upgrade to 1.2.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.4-1" date="1599220800">- Upgrade to 1.2.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.5-1" date="1600430400">- Upgrade to 1.2.5</changelog>
</package>
<package pkgid="393875e67e0d94c4c9988d9b0944e4c93daa6dfb2dd7065ffcbfe2f273b34686" name="freetds" arch="src">
  <version epoch="0" ver="1.2.6" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.39-1" date="1589716800">- Upgrade to 1.1.39</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.40-1" date="1590148800">- Upgrade to 1.1.40</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.41-1" date="1592049600">- Upgrade to 1.1.41</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.42-1" date="1592136000">- Upgrade to 1.1.42</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2-1" date="1593777600">- Upgrade to 1.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.2-1" date="1594123200">- Upgrade to 1.2.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.3-1" date="1594296000">- Upgrade to 1.2.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.4-1" date="1599220800">- Upgrade to 1.2.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.5-1" date="1600430400">- Upgrade to 1.2.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.6-1" date="1603972800">- Upgrade to 1.2.6</changelog>
</package>
<package pkgid="629d78e96510fea7e1a32ee0ef4ebf5ecf9909a0eb372bd1e3e572c6b44e5248" name="freetds" arch="src">
  <version epoch="0" ver="1.2.9" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.40-1" date="1590148800">- Upgrade to 1.1.40</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.41-1" date="1592049600">- Upgrade to 1.1.41</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.42-1" date="1592136000">- Upgrade to 1.1.42</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2-1" date="1593777600">- Upgrade to 1.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.2-1" date="1594123200">- Upgrade to 1.2.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.3-1" date="1594296000">- Upgrade to 1.2.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.4-1" date="1599220800">- Upgrade to 1.2.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.5-1" date="1600430400">- Upgrade to 1.2.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.6-1" date="1603972800">- Upgrade to 1.2.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.9-1" date="1604059200">- Upgrade to 1.2.9</changelog>
</package>
<package pkgid="3b36de715f713f5c578ffe5cbbd57cfd9d630c6ea4bc0309c3ccae8bdf14e93c" name="freetds" arch="src">
  <version epoch="0" ver="1.3" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.10-1" date="1604145600">- Upgrade to 1.2.10</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.11-1" date="1605182400">- Upgrade to 1.2.11</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.12-1" date="1605960000">- Upgrade to 1.2.12</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.13-1" date="1606651200">- Upgrade to 1.2.13</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.17-1" date="1607515200">- Upgrade to 1.2.17</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.18-1" date="1608292800">- Upgrade to 1.2.18</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.19-1" date="1618056000">- Upgrade to 1.2.19</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.20-1" date="1618315200">- Upgrade to 1.2.20</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.21-1" date="1619265600">- Upgrade to 1.2.21</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3-1" date="1626177600">- Upgrade to 1.3</changelog>
</package>
<package pkgid="ef616858560699e4280e1ef98a14547f6a959272eaccc47107fb513217aa9e89" name="freetds" arch="src">
  <version epoch="0" ver="1.3.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.11-1" date="1605182400">- Upgrade to 1.2.11</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.12-1" date="1605960000">- Upgrade to 1.2.12</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.13-1" date="1606651200">- Upgrade to 1.2.13</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.17-1" date="1607515200">- Upgrade to 1.2.17</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.18-1" date="1608292800">- Upgrade to 1.2.18</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.19-1" date="1618056000">- Upgrade to 1.2.19</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.20-1" date="1618315200">- Upgrade to 1.2.20</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.21-1" date="1619265600">- Upgrade to 1.2.21</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3-1" date="1626177600">- Upgrade to 1.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.1-1" date="1629288000">- Upgrade to 1.3.1</changelog>
</package>
<package pkgid="07ebcd634a456b2b6dbc1f3e978112e4ad1cf924e2c9a9b772ffb7202a2ebe8f" name="freetds" arch="src">
  <version epoch="0" ver="1.3.10" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.1-1" date="1629288000">- Upgrade to 1.3.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.2-1" date="1629892800">- Upgrade to 1.3.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.3-1" date="1630411200">- Upgrade to 1.3.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.4-1" date="1639396800">- Upgrade to 1.3.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.5-1" date="1640433600">- Upgrade to 1.3.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.6-1" date="1640606400">- Upgrade to 1.3.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.7-1" date="1644062400">- Upgrade to 1.3.7</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.8-1" date="1644494400">- Upgrade to 1.3.8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.9-1" date="1644580800">- Upgrade to 1.3.9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.10-1" date="1650024000">- Upgrade to 1.3.10</changelog>
</package>
<package pkgid="3e37c941c2fd73b93949a1b3d166330d97243a04f9b37ecf6e5ce02890cc8106" name="freetds" arch="src">
  <version epoch="0" ver="1.3.11" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.2-1" date="1629892800">- Upgrade to 1.3.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.3-1" date="1630411200">- Upgrade to 1.3.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.4-1" date="1639396800">- Upgrade to 1.3.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.5-1" date="1640433600">- Upgrade to 1.3.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.6-1" date="1640606400">- Upgrade to 1.3.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.7-1" date="1644062400">- Upgrade to 1.3.7</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.8-1" date="1644494400">- Upgrade to 1.3.8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.9-1" date="1644580800">- Upgrade to 1.3.9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.10-1" date="1650024000">- Upgrade to 1.3.10</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.11-1" date="1653998400">- Upgrade to 1.3.11</changelog>
</package>
<package pkgid="c83865f130be30c84ab9e901a51501d61fc1141126e63e5edd68c6e7d9f440e3" name="freetds" arch="src">
  <version epoch="0" ver="1.3.12" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.3-1" date="1630411200">- Upgrade to 1.3.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.4-1" date="1639396800">- Upgrade to 1.3.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.5-1" date="1640433600">- Upgrade to 1.3.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.6-1" date="1640606400">- Upgrade to 1.3.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.7-1" date="1644062400">- Upgrade to 1.3.7</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.8-1" date="1644494400">- Upgrade to 1.3.8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.9-1" date="1644580800">- Upgrade to 1.3.9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.10-1" date="1650024000">- Upgrade to 1.3.10</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.11-1" date="1653998400">- Upgrade to 1.3.11</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.12-1" date="1654948800">- Upgrade to 1.3.12</changelog>
</package>
<package pkgid="6bc9126771294fdd33f737a69df3702eb252d9009c2bd50d66b4181fe1b9e15f" name="freetds" arch="src">
  <version epoch="0" ver="1.3.13" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.4-1" date="1639396800">- Upgrade to 1.3.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.5-1" date="1640433600">- Upgrade to 1.3.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.6-1" date="1640606400">- Upgrade to 1.3.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.7-1" date="1644062400">- Upgrade to 1.3.7</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.8-1" date="1644494400">- Upgrade to 1.3.8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.9-1" date="1644580800">- Upgrade to 1.3.9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.10-1" date="1650024000">- Upgrade to 1.3.10</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.11-1" date="1653998400">- Upgrade to 1.3.11</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.12-1" date="1654948800">- Upgrade to 1.3.12</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.13-1" date="1660737600">- Upgrade to 1.3.13</changelog>
</package>
<package pkgid="f2bcd67f997556a1e33e4bd830e4b9e05b2cd556f0fd540f3e2251cd0b61fa09" name="freetds" arch="src">
  <version epoch="0" ver="1.3.14" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.5-1" date="1640433600">- Upgrade to 1.3.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.6-1" date="1640606400">- Upgrade to 1.3.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.7-1" date="1644062400">- Upgrade to 1.3.7</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.8-1" date="1644494400">- Upgrade to 1.3.8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.9-1" date="1644580800">- Upgrade to 1.3.9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.10-1" date="1650024000">- Upgrade to 1.3.10</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.11-1" date="1653998400">- Upgrade to 1.3.11</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.12-1" date="1654948800">- Upgrade to 1.3.12</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.13-1" date="1660737600">- Upgrade to 1.3.13</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.14-1" date="1667908800">- Upgrade to 1.3.14</changelog>
</package>
<package pkgid="4d78c90291c06de2f29cf4cde376472cb417544238f43810e9d463bf7982c069" name="freetds" arch="src">
  <version epoch="0" ver="1.3.15" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.6-1" date="1640606400">- Upgrade to 1.3.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.7-1" date="1644062400">- Upgrade to 1.3.7</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.8-1" date="1644494400">- Upgrade to 1.3.8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.9-1" date="1644580800">- Upgrade to 1.3.9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.10-1" date="1650024000">- Upgrade to 1.3.10</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.11-1" date="1653998400">- Upgrade to 1.3.11</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.12-1" date="1654948800">- Upgrade to 1.3.12</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.13-1" date="1660737600">- Upgrade to 1.3.13</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.14-1" date="1667908800">- Upgrade to 1.3.14</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.15-1" date="1669118400">- Upgrade to 1.3.15</changelog>
</package>
<package pkgid="fe4ed5144f9122eef7fd4a74e654bbdb0db97a15d11444b819c0743a7d475685" name="freetds" arch="src">
  <version epoch="0" ver="1.3.16" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.7-1" date="1644062400">- Upgrade to 1.3.7</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.8-1" date="1644494400">- Upgrade to 1.3.8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.9-1" date="1644580800">- Upgrade to 1.3.9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.10-1" date="1650024000">- Upgrade to 1.3.10</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.11-1" date="1653998400">- Upgrade to 1.3.11</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.12-1" date="1654948800">- Upgrade to 1.3.12</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.13-1" date="1660737600">- Upgrade to 1.3.13</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.14-1" date="1667908800">- Upgrade to 1.3.14</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.15-1" date="1669118400">- Upgrade to 1.3.15</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.16-1" date="1670587200">- Upgrade to 1.3.16</changelog>
</package>
<package pkgid="99f25aa72bf32c69a3336da0c03b20e7bc489d0855dc5a22b42bbc0ea85308d3" name="freetds" arch="src">
  <version epoch="0" ver="1.3.17" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.8-1" date="1644494400">- Upgrade to 1.3.8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.9-1" date="1644580800">- Upgrade to 1.3.9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.10-1" date="1650024000">- Upgrade to 1.3.10</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.11-1" date="1653998400">- Upgrade to 1.3.11</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.12-1" date="1654948800">- Upgrade to 1.3.12</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.13-1" date="1660737600">- Upgrade to 1.3.13</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.14-1" date="1667908800">- Upgrade to 1.3.14</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.15-1" date="1669118400">- Upgrade to 1.3.15</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.16-1" date="1670587200">- Upgrade to 1.3.16</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.17-1" date="1674302400">- Upgrade to 1.3.17</changelog>
</package>
<package pkgid="b6bfaece4d81925ed24fa178b08593ab3a7d85c7e9c4a511c6fdfcdfa9a536ec" name="freetds" arch="src">
  <version epoch="0" ver="1.3.18" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.9-1" date="1644580800">- Upgrade to 1.3.9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.10-1" date="1650024000">- Upgrade to 1.3.10</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.11-1" date="1653998400">- Upgrade to 1.3.11</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.12-1" date="1654948800">- Upgrade to 1.3.12</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.13-1" date="1660737600">- Upgrade to 1.3.13</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.14-1" date="1667908800">- Upgrade to 1.3.14</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.15-1" date="1669118400">- Upgrade to 1.3.15</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.16-1" date="1670587200">- Upgrade to 1.3.16</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.17-1" date="1674302400">- Upgrade to 1.3.17</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.18-1" date="1680350400">- Upgrade to 1.3.18</changelog>
</package>
<package pkgid="fda023e4cdc1c17a90639136f5b4d255048604238389a8c727a121ed4c7f6bf7" name="freetds" arch="src">
  <version epoch="0" ver="1.3.19" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.10-1" date="1650024000">- Upgrade to 1.3.10</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.11-1" date="1653998400">- Upgrade to 1.3.11</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.12-1" date="1654948800">- Upgrade to 1.3.12</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.13-1" date="1660737600">- Upgrade to 1.3.13</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.14-1" date="1667908800">- Upgrade to 1.3.14</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.15-1" date="1669118400">- Upgrade to 1.3.15</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.16-1" date="1670587200">- Upgrade to 1.3.16</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.17-1" date="1674302400">- Upgrade to 1.3.17</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.18-1" date="1680350400">- Upgrade to 1.3.18</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.19-1" date="1692014400">- Upgrade to 1.3.19</changelog>
</package>
<package pkgid="d14a07fb19ca4a3ffdb165bcf6e8925ff9fc57921a994095d5087c7f4ff2c271" name="freetds" arch="src">
  <version epoch="0" ver="1.3.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.12-1" date="1605960000">- Upgrade to 1.2.12</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.13-1" date="1606651200">- Upgrade to 1.2.13</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.17-1" date="1607515200">- Upgrade to 1.2.17</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.18-1" date="1608292800">- Upgrade to 1.2.18</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.19-1" date="1618056000">- Upgrade to 1.2.19</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.20-1" date="1618315200">- Upgrade to 1.2.20</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.21-1" date="1619265600">- Upgrade to 1.2.21</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3-1" date="1626177600">- Upgrade to 1.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.1-1" date="1629288000">- Upgrade to 1.3.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.2-1" date="1629892800">- Upgrade to 1.3.2</changelog>
</package>
<package pkgid="0aba89ae7be00534a995e09526a7ce89e0d7e93d3074ebe562f89e141a2586fa" name="freetds" arch="src">
  <version epoch="0" ver="1.3.20" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.11-1" date="1653998400">- Upgrade to 1.3.11</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.12-1" date="1654948800">- Upgrade to 1.3.12</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.13-1" date="1660737600">- Upgrade to 1.3.13</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.14-1" date="1667908800">- Upgrade to 1.3.14</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.15-1" date="1669118400">- Upgrade to 1.3.15</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.16-1" date="1670587200">- Upgrade to 1.3.16</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.17-1" date="1674302400">- Upgrade to 1.3.17</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.18-1" date="1680350400">- Upgrade to 1.3.18</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.19-1" date="1692014400">- Upgrade to 1.3.19</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.20-1" date="1693137600">- Upgrade to 1.3.20</changelog>
</package>
<package pkgid="d2a26f3ca6d605dabb56ec54ef227775257f30fbf95a3da9cb29903453c1a251" name="freetds" arch="src">
  <version epoch="0" ver="1.3.21" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.15-1" date="1669118400">- Upgrade to 1.3.15</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.16-1" date="1670587200">- Upgrade to 1.3.16</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.17-1" date="1674302400">- Upgrade to 1.3.17</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.18-1" date="1680350400">- Upgrade to 1.3.18</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.19-1" date="1692014400">- Upgrade to 1.3.19</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.20-1" date="1693137600">- Upgrade to 1.3.20</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4-1" date="1695643200">- Upgrade to 1.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.1-1" date="1695729600">- Upgrade to 1.4.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.2-1" date="1695729601">- Upgrade to 1.4.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.21-1" date="1695816000">- Upgrade to 1.3.21</changelog>
</package>
<package pkgid="d468ecfa9ad712ccc91dafcd57829625d2d0e4e31f6e958ab5892453c05dc48c" name="freetds" arch="src">
  <version epoch="0" ver="1.3.3" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.13-1" date="1606651200">- Upgrade to 1.2.13</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.17-1" date="1607515200">- Upgrade to 1.2.17</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.18-1" date="1608292800">- Upgrade to 1.2.18</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.19-1" date="1618056000">- Upgrade to 1.2.19</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.20-1" date="1618315200">- Upgrade to 1.2.20</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.21-1" date="1619265600">- Upgrade to 1.2.21</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3-1" date="1626177600">- Upgrade to 1.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.1-1" date="1629288000">- Upgrade to 1.3.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.2-1" date="1629892800">- Upgrade to 1.3.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.3-1" date="1630411200">- Upgrade to 1.3.3</changelog>
</package>
<package pkgid="963385f2bffac7c145dfba13c8853aed50ebbef50c815b9fb638ac93cabc59f2" name="freetds" arch="src">
  <version epoch="0" ver="1.3.4" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.17-1" date="1607515200">- Upgrade to 1.2.17</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.18-1" date="1608292800">- Upgrade to 1.2.18</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.19-1" date="1618056000">- Upgrade to 1.2.19</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.20-1" date="1618315200">- Upgrade to 1.2.20</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.21-1" date="1619265600">- Upgrade to 1.2.21</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3-1" date="1626177600">- Upgrade to 1.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.1-1" date="1629288000">- Upgrade to 1.3.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.2-1" date="1629892800">- Upgrade to 1.3.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.3-1" date="1630411200">- Upgrade to 1.3.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.4-1" date="1639396800">- Upgrade to 1.3.4</changelog>
</package>
<package pkgid="8d1c5304e62b45b434f0024d99164fec90e89190f70816023dd03d1ff9932197" name="freetds" arch="src">
  <version epoch="0" ver="1.3.5" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.18-1" date="1608292800">- Upgrade to 1.2.18</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.19-1" date="1618056000">- Upgrade to 1.2.19</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.20-1" date="1618315200">- Upgrade to 1.2.20</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.21-1" date="1619265600">- Upgrade to 1.2.21</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3-1" date="1626177600">- Upgrade to 1.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.1-1" date="1629288000">- Upgrade to 1.3.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.2-1" date="1629892800">- Upgrade to 1.3.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.3-1" date="1630411200">- Upgrade to 1.3.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.4-1" date="1639396800">- Upgrade to 1.3.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.5-1" date="1640433600">- Upgrade to 1.3.5</changelog>
</package>
<package pkgid="cc81fc7ac88591d8f0afac9a61ca8daf4d49f8676702845846357c32ec94051e" name="freetds" arch="src">
  <version epoch="0" ver="1.3.6" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.19-1" date="1618056000">- Upgrade to 1.2.19</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.20-1" date="1618315200">- Upgrade to 1.2.20</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.21-1" date="1619265600">- Upgrade to 1.2.21</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3-1" date="1626177600">- Upgrade to 1.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.1-1" date="1629288000">- Upgrade to 1.3.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.2-1" date="1629892800">- Upgrade to 1.3.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.3-1" date="1630411200">- Upgrade to 1.3.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.4-1" date="1639396800">- Upgrade to 1.3.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.5-1" date="1640433600">- Upgrade to 1.3.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.6-1" date="1640606400">- Upgrade to 1.3.6</changelog>
</package>
<package pkgid="49a06b0789e4ab2d67f5223a0a48a6f79c22adbe8faa4cbd1f3d9f6aedc6b8aa" name="freetds" arch="src">
  <version epoch="0" ver="1.3.7" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.20-1" date="1618315200">- Upgrade to 1.2.20</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.21-1" date="1619265600">- Upgrade to 1.2.21</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3-1" date="1626177600">- Upgrade to 1.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.1-1" date="1629288000">- Upgrade to 1.3.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.2-1" date="1629892800">- Upgrade to 1.3.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.3-1" date="1630411200">- Upgrade to 1.3.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.4-1" date="1639396800">- Upgrade to 1.3.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.5-1" date="1640433600">- Upgrade to 1.3.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.6-1" date="1640606400">- Upgrade to 1.3.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.7-1" date="1644062400">- Upgrade to 1.3.7</changelog>
</package>
<package pkgid="bb86006d0e4343a07fca997102b20cd45318b8d9e5a3d79788dfca5f4118b475" name="freetds" arch="src">
  <version epoch="0" ver="1.3.8" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.21-1" date="1619265600">- Upgrade to 1.2.21</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3-1" date="1626177600">- Upgrade to 1.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.1-1" date="1629288000">- Upgrade to 1.3.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.2-1" date="1629892800">- Upgrade to 1.3.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.3-1" date="1630411200">- Upgrade to 1.3.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.4-1" date="1639396800">- Upgrade to 1.3.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.5-1" date="1640433600">- Upgrade to 1.3.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.6-1" date="1640606400">- Upgrade to 1.3.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.7-1" date="1644062400">- Upgrade to 1.3.7</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.8-1" date="1644494400">- Upgrade to 1.3.8</changelog>
</package>
<package pkgid="d373f08f44b386d994052635df6be4246b9e36e79fa81f7f3cab3866557cf364" name="freetds" arch="src">
  <version epoch="0" ver="1.3.9" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3-1" date="1626177600">- Upgrade to 1.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.1-1" date="1629288000">- Upgrade to 1.3.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.2-1" date="1629892800">- Upgrade to 1.3.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.3-1" date="1630411200">- Upgrade to 1.3.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.4-1" date="1639396800">- Upgrade to 1.3.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.5-1" date="1640433600">- Upgrade to 1.3.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.6-1" date="1640606400">- Upgrade to 1.3.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.7-1" date="1644062400">- Upgrade to 1.3.7</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.8-1" date="1644494400">- Upgrade to 1.3.8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.9-1" date="1644580800">- Upgrade to 1.3.9</changelog>
</package>
<package pkgid="4ef73debf0d2dc97374513d4d090edf4c46e1cbf5ea151930a87a11aec79b2a1" name="freetds" arch="src">
  <version epoch="0" ver="1.4" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.12-1" date="1654948800">- Upgrade to 1.3.12</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.13-1" date="1660737600">- Upgrade to 1.3.13</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.14-1" date="1667908800">- Upgrade to 1.3.14</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.15-1" date="1669118400">- Upgrade to 1.3.15</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.16-1" date="1670587200">- Upgrade to 1.3.16</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.17-1" date="1674302400">- Upgrade to 1.3.17</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.18-1" date="1680350400">- Upgrade to 1.3.18</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.19-1" date="1692014400">- Upgrade to 1.3.19</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.20-1" date="1693137600">- Upgrade to 1.3.20</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4-1" date="1695643200">- Upgrade to 1.4</changelog>
</package>
<package pkgid="4093ca7b3a0c79c8a262ede92c97eb72f67bf4a5ac0077af45978a9a4ebf45e9" name="freetds" arch="src">
  <version epoch="0" ver="1.4.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.13-1" date="1660737600">- Upgrade to 1.3.13</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.14-1" date="1667908800">- Upgrade to 1.3.14</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.15-1" date="1669118400">- Upgrade to 1.3.15</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.16-1" date="1670587200">- Upgrade to 1.3.16</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.17-1" date="1674302400">- Upgrade to 1.3.17</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.18-1" date="1680350400">- Upgrade to 1.3.18</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.19-1" date="1692014400">- Upgrade to 1.3.19</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.20-1" date="1693137600">- Upgrade to 1.3.20</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4-1" date="1695643200">- Upgrade to 1.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.1-1" date="1695729600">- Upgrade to 1.4.1</changelog>
</package>
<package pkgid="434462b103f8f926c60056b0c6f5e5681c299f0f4bf58950a97c95982c2ac901" name="freetds" arch="src">
  <version epoch="0" ver="1.4.10" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.2-1" date="1695729600">- Upgrade to 1.4.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.21-1" date="1695816000">- Upgrade to 1.3.21</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.3-1" date="1697198400">- Upgrade to 1.4.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.4-1" date="1697803200">- Upgrade to 1.4.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.5-1" date="1697976000">- Upgrade to 1.4.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.6-1" date="1698408000">- Upgrade to 1.4.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.7-1" date="1700827200">- Upgrade to 1.4.7</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.8-1" date="1701000000">- Upgrade to 1.4.8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.9-1" date="1701604800">- Upgrade to 1.4.9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.10-1" date="1701691200">- Upgrade to 1.4.10</changelog>
</package>
<package pkgid="5b1908eb5b3b62792d80072233bd40d32616c88a142f03f6d9c9ed64d4a86d1d" name="freetds" arch="src">
  <version epoch="0" ver="1.4.11" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.21-1" date="1695816000">- Upgrade to 1.3.21</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.3-1" date="1697198400">- Upgrade to 1.4.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.4-1" date="1697803200">- Upgrade to 1.4.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.5-1" date="1697976000">- Upgrade to 1.4.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.6-1" date="1698408000">- Upgrade to 1.4.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.7-1" date="1700827200">- Upgrade to 1.4.7</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.8-1" date="1701000000">- Upgrade to 1.4.8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.9-1" date="1701604800">- Upgrade to 1.4.9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.10-1" date="1701691200">- Upgrade to 1.4.10</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.11-1" date="1711713600">- Upgrade to 1.4.11</changelog>
</package>
<package pkgid="6f03ba76c84d80cd37c92c60bdd83183b7e15bab22249670088d69cd96f6ebe8" name="freetds" arch="src">
  <version epoch="0" ver="1.4.12" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.3-1" date="1697198400">- Upgrade to 1.4.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.4-1" date="1697803200">- Upgrade to 1.4.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.5-1" date="1697976000">- Upgrade to 1.4.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.6-1" date="1698408000">- Upgrade to 1.4.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.7-1" date="1700827200">- Upgrade to 1.4.7</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.8-1" date="1701000000">- Upgrade to 1.4.8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.9-1" date="1701604800">- Upgrade to 1.4.9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.10-1" date="1701691200">- Upgrade to 1.4.10</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.11-1" date="1711713600">- Upgrade to 1.4.11</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.12-1" date="1711972800">- Upgrade to 1.4.12</changelog>
</package>
<package pkgid="ce9e256c062d943088193fea015721f865263cd45b0e99573cbd3965944b1409" name="freetds" arch="src">
  <version epoch="0" ver="1.4.13" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.4-1" date="1697803200">- Upgrade to 1.4.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.5-1" date="1697976000">- Upgrade to 1.4.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.6-1" date="1698408000">- Upgrade to 1.4.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.7-1" date="1700827200">- Upgrade to 1.4.7</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.8-1" date="1701000000">- Upgrade to 1.4.8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.9-1" date="1701604800">- Upgrade to 1.4.9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.10-1" date="1701691200">- Upgrade to 1.4.10</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.11-1" date="1711713600">- Upgrade to 1.4.11</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.12-1" date="1711972800">- Upgrade to 1.4.12</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.13-1" date="1715256000">- Upgrade to 1.4.13</changelog>
</package>
<package pkgid="9b03c956a41735bef80f9900c42fe806fc652cec56473dd3220bb88c5cf8ec0c" name="freetds" arch="src">
  <version epoch="0" ver="1.4.14" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.5-1" date="1697976000">- Upgrade to 1.4.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.6-1" date="1698408000">- Upgrade to 1.4.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.7-1" date="1700827200">- Upgrade to 1.4.7</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.8-1" date="1701000000">- Upgrade to 1.4.8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.9-1" date="1701604800">- Upgrade to 1.4.9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.10-1" date="1701691200">- Upgrade to 1.4.10</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.11-1" date="1711713600">- Upgrade to 1.4.11</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.12-1" date="1711972800">- Upgrade to 1.4.12</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.13-1" date="1715256000">- Upgrade to 1.4.13</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.14-1" date="1715947200">- Upgrade to 1.4.14</changelog>
</package>
<package pkgid="d9bf85f78147787728a65043b8ea87506069d256fd316ceef8ca1e86f1732b17" name="freetds" arch="src">
  <version epoch="0" ver="1.4.15" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.6-1" date="1698408000">- Upgrade to 1.4.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.7-1" date="1700827200">- Upgrade to 1.4.7</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.8-1" date="1701000000">- Upgrade to 1.4.8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.9-1" date="1701604800">- Upgrade to 1.4.9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.10-1" date="1701691200">- Upgrade to 1.4.10</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.11-1" date="1711713600">- Upgrade to 1.4.11</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.12-1" date="1711972800">- Upgrade to 1.4.12</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.13-1" date="1715256000">- Upgrade to 1.4.13</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.14-1" date="1715947200">- Upgrade to 1.4.14</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.15-1" date="1716897600">- Upgrade to 1.4.15</changelog>
</package>
<package pkgid="941fc4748ee0b850b1095e7a462b1efbb49b2bce7f5117ea927648feec502dec" name="freetds" arch="src">
  <version epoch="0" ver="1.4.16" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.7-1" date="1700827200">- Upgrade to 1.4.7</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.8-1" date="1701000000">- Upgrade to 1.4.8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.9-1" date="1701604800">- Upgrade to 1.4.9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.10-1" date="1701691200">- Upgrade to 1.4.10</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.11-1" date="1711713600">- Upgrade to 1.4.11</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.12-1" date="1711972800">- Upgrade to 1.4.12</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.13-1" date="1715256000">- Upgrade to 1.4.13</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.14-1" date="1715947200">- Upgrade to 1.4.14</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.15-1" date="1716897600">- Upgrade to 1.4.15</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.16-1" date="1716984000">- Upgrade to 1.4.16</changelog>
</package>
<package pkgid="312dbfdf193e6af40c6d6113bad427f27dc82c116b4a9c50f62bfed3c3b6e060" name="freetds" arch="src">
  <version epoch="0" ver="1.4.17" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.8-1" date="1701000000">- Upgrade to 1.4.8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.9-1" date="1701604800">- Upgrade to 1.4.9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.10-1" date="1701691200">- Upgrade to 1.4.10</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.11-1" date="1711713600">- Upgrade to 1.4.11</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.12-1" date="1711972800">- Upgrade to 1.4.12</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.13-1" date="1715256000">- Upgrade to 1.4.13</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.14-1" date="1715947200">- Upgrade to 1.4.14</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.15-1" date="1716897600">- Upgrade to 1.4.15</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.16-1" date="1716984000">- Upgrade to 1.4.16</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.17-1" date="1717416000">- Upgrade to 1.4.17</changelog>
</package>
<package pkgid="72c4896dd4128e7a3b9a8f3110604bcd5a63f796d815b645e0fc245784353870" name="freetds" arch="src">
  <version epoch="0" ver="1.4.19" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.9-1" date="1701604800">- Upgrade to 1.4.9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.10-1" date="1701691200">- Upgrade to 1.4.10</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.11-1" date="1711713600">- Upgrade to 1.4.11</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.12-1" date="1711972800">- Upgrade to 1.4.12</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.13-1" date="1715256000">- Upgrade to 1.4.13</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.14-1" date="1715947200">- Upgrade to 1.4.14</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.15-1" date="1716897600">- Upgrade to 1.4.15</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.16-1" date="1716984000">- Upgrade to 1.4.16</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.17-1" date="1717416000">- Upgrade to 1.4.17</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.19-1" date="1720094400">- Upgrade to 1.4.19</changelog>
</package>
<package pkgid="be1fc4e3764bd8e2da2c319ff4dda36543245699fa0b2ad95fe49e3c2b17b517" name="freetds" arch="src">
  <version epoch="0" ver="1.4.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.14-1" date="1667908800">- Upgrade to 1.3.14</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.15-1" date="1669118400">- Upgrade to 1.3.15</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.16-1" date="1670587200">- Upgrade to 1.3.16</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.17-1" date="1674302400">- Upgrade to 1.3.17</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.18-1" date="1680350400">- Upgrade to 1.3.18</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.19-1" date="1692014400">- Upgrade to 1.3.19</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.20-1" date="1693137600">- Upgrade to 1.3.20</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4-1" date="1695643200">- Upgrade to 1.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.1-1" date="1695729600">- Upgrade to 1.4.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.2-1" date="1695729601">- Upgrade to 1.4.2</changelog>
</package>
<package pkgid="b7ee8983253e3e630ad41467e410bf7ebc5fdeec9d6560a21c1e673c87a500e0" name="freetds" arch="src">
  <version epoch="0" ver="1.4.20" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.10-1" date="1701691200">- Upgrade to 1.4.10</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.11-1" date="1711713600">- Upgrade to 1.4.11</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.12-1" date="1711972800">- Upgrade to 1.4.12</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.13-1" date="1715256000">- Upgrade to 1.4.13</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.14-1" date="1715947200">- Upgrade to 1.4.14</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.15-1" date="1716897600">- Upgrade to 1.4.15</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.16-1" date="1716984000">- Upgrade to 1.4.16</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.17-1" date="1717416000">- Upgrade to 1.4.17</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.19-1" date="1720094400">- Upgrade to 1.4.19</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.20-1" date="1721131200">- Upgrade to 1.4.20</changelog>
</package>
<package pkgid="b6efd98b822ca9394033bb86c23936001ae0c7764d6cab181029b5607c8eb0ad" name="freetds" arch="src">
  <version epoch="0" ver="1.4.21" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.11-1" date="1711713600">- Upgrade to 1.4.11</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.12-1" date="1711972800">- Upgrade to 1.4.12</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.13-1" date="1715256000">- Upgrade to 1.4.13</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.14-1" date="1715947200">- Upgrade to 1.4.14</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.15-1" date="1716897600">- Upgrade to 1.4.15</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.16-1" date="1716984000">- Upgrade to 1.4.16</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.17-1" date="1717416000">- Upgrade to 1.4.17</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.19-1" date="1720094400">- Upgrade to 1.4.19</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.20-1" date="1721131200">- Upgrade to 1.4.20</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.21-1" date="1721563200">- Upgrade to 1.4.21</changelog>
</package>
<package pkgid="e12c6e0f3b09544a44f48d6673402daea0a472ab022906688580c28472a22677" name="freetds" arch="src">
  <version epoch="0" ver="1.4.22" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.12-1" date="1711972800">- Upgrade to 1.4.12</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.13-1" date="1715256000">- Upgrade to 1.4.13</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.14-1" date="1715947200">- Upgrade to 1.4.14</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.15-1" date="1716897600">- Upgrade to 1.4.15</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.16-1" date="1716984000">- Upgrade to 1.4.16</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.17-1" date="1717416000">- Upgrade to 1.4.17</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.19-1" date="1720094400">- Upgrade to 1.4.19</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.20-1" date="1721131200">- Upgrade to 1.4.20</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.21-1" date="1721563200">- Upgrade to 1.4.21</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.22-1" date="1721822400">- Upgrade to 1.4.22</changelog>
</package>
<package pkgid="0751d3fcd25ebda593b570282f6b2d163a59a3ea34c77b485bc5df8a41326216" name="freetds" arch="src">
  <version epoch="0" ver="1.4.3" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.16-1" date="1670587200">- Upgrade to 1.3.16</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.17-1" date="1674302400">- Upgrade to 1.3.17</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.18-1" date="1680350400">- Upgrade to 1.3.18</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.19-1" date="1692014400">- Upgrade to 1.3.19</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.20-1" date="1693137600">- Upgrade to 1.3.20</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4-1" date="1695643200">- Upgrade to 1.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.1-1" date="1695729600">- Upgrade to 1.4.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.2-1" date="1695729601">- Upgrade to 1.4.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.21-1" date="1695816000">- Upgrade to 1.3.21</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.3-1" date="1697198400">- Upgrade to 1.4.3</changelog>
</package>
<package pkgid="e0a675623e0e4a15911c7eaf42c78757a4810595356b6df4548eccddc7425746" name="freetds" arch="src">
  <version epoch="0" ver="1.4.4" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.17-1" date="1674302400">- Upgrade to 1.3.17</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.18-1" date="1680350400">- Upgrade to 1.3.18</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.19-1" date="1692014400">- Upgrade to 1.3.19</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.20-1" date="1693137600">- Upgrade to 1.3.20</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4-1" date="1695643200">- Upgrade to 1.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.1-1" date="1695729600">- Upgrade to 1.4.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.2-1" date="1695729601">- Upgrade to 1.4.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.21-1" date="1695816000">- Upgrade to 1.3.21</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.3-1" date="1697198400">- Upgrade to 1.4.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.4-1" date="1697803200">- Upgrade to 1.4.4</changelog>
</package>
<package pkgid="617b1255abdbdf7fa3270178e928e3c0c822c5164f901a35b8714e9440d2c0ad" name="freetds" arch="src">
  <version epoch="0" ver="1.4.5" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.18-1" date="1680350400">- Upgrade to 1.3.18</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.19-1" date="1692014400">- Upgrade to 1.3.19</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.20-1" date="1693137600">- Upgrade to 1.3.20</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4-1" date="1695643200">- Upgrade to 1.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.1-1" date="1695729600">- Upgrade to 1.4.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.2-1" date="1695729601">- Upgrade to 1.4.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.21-1" date="1695816000">- Upgrade to 1.3.21</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.3-1" date="1697198400">- Upgrade to 1.4.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.4-1" date="1697803200">- Upgrade to 1.4.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.5-1" date="1697976000">- Upgrade to 1.4.5</changelog>
</package>
<package pkgid="d268633bbb49b17674f8e33831a43e60ebf1a787b8c2c5a742020142c0d4611a" name="freetds" arch="src">
  <version epoch="0" ver="1.4.6" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.19-1" date="1692014400">- Upgrade to 1.3.19</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.20-1" date="1693137600">- Upgrade to 1.3.20</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4-1" date="1695643200">- Upgrade to 1.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.1-1" date="1695729600">- Upgrade to 1.4.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.2-1" date="1695729601">- Upgrade to 1.4.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.21-1" date="1695816000">- Upgrade to 1.3.21</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.3-1" date="1697198400">- Upgrade to 1.4.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.4-1" date="1697803200">- Upgrade to 1.4.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.5-1" date="1697976000">- Upgrade to 1.4.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.6-1" date="1698408000">- Upgrade to 1.4.6</changelog>
</package>
<package pkgid="c02df74116f2c2658ba9adcda75f2bf593d5bfa95e699e881c6c88052218a3b9" name="freetds" arch="src">
  <version epoch="0" ver="1.4.7" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.20-1" date="1693137600">- Upgrade to 1.3.20</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4-1" date="1695643200">- Upgrade to 1.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.1-1" date="1695729600">- Upgrade to 1.4.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.2-1" date="1695729601">- Upgrade to 1.4.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.21-1" date="1695816000">- Upgrade to 1.3.21</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.3-1" date="1697198400">- Upgrade to 1.4.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.4-1" date="1697803200">- Upgrade to 1.4.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.5-1" date="1697976000">- Upgrade to 1.4.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.6-1" date="1698408000">- Upgrade to 1.4.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.7-1" date="1700827200">- Upgrade to 1.4.7</changelog>
</package>
<package pkgid="4aa9db8d9b60a8b19ae9f35e9d7eea2fcba80f2ba0639c02c28909be201eed20" name="freetds" arch="src">
  <version epoch="0" ver="1.4.8" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4-1" date="1695643200">- Upgrade to 1.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.1-1" date="1695729600">- Upgrade to 1.4.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.2-1" date="1695729601">- Upgrade to 1.4.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.21-1" date="1695816000">- Upgrade to 1.3.21</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.3-1" date="1697198400">- Upgrade to 1.4.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.4-1" date="1697803200">- Upgrade to 1.4.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.5-1" date="1697976000">- Upgrade to 1.4.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.6-1" date="1698408000">- Upgrade to 1.4.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.7-1" date="1700827200">- Upgrade to 1.4.7</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.8-1" date="1701000000">- Upgrade to 1.4.8</changelog>
</package>
<package pkgid="0d23d202c7568f89e6be354965bbdfbf4a26831c73f6d8e8b2a9d11549efa341" name="freetds" arch="src">
  <version epoch="0" ver="1.4.9" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.1-1" date="1695729600">- Upgrade to 1.4.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.2-1" date="1695729601">- Upgrade to 1.4.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.21-1" date="1695816000">- Upgrade to 1.3.21</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.3-1" date="1697198400">- Upgrade to 1.4.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.4-1" date="1697803200">- Upgrade to 1.4.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.5-1" date="1697976000">- Upgrade to 1.4.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.6-1" date="1698408000">- Upgrade to 1.4.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.7-1" date="1700827200">- Upgrade to 1.4.7</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.8-1" date="1701000000">- Upgrade to 1.4.8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.9-1" date="1701604800">- Upgrade to 1.4.9</changelog>
</package>
<package pkgid="fb8255509e8bda567f62881e56ee60af4bd72290042a08fb20aa914225425fb6" name="gearmand" arch="src">
  <version epoch="0" ver="1.1.19.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.13-1" date="1600344000">- Update to release 1.1.13</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.14-1" date="1600344001">- Update to release 1.1.14</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.15-1" date="1600344002">- Update to release 1.1.15</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.16-1" date="1600344003">- Update to release 1.1.16</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.17-1" date="1600344004">- Update to release 1.1.17</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.18-1" date="1600344005">- Update to release 1.1.18</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.19-1" date="1600344006">- Update to release 1.1.19</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.19.1-1" date="1600344007">- Update to release 1.1.19.1</changelog>
</package>
<package pkgid="84343b98fd214f45715eba35d1788845a99e39d41b8e9a0f8637328984978e06" name="gearmand" arch="src">
  <version epoch="0" ver="1.1.20" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.20-1" date="1675771200">- Update to release 1.1.20</changelog>
</package>
<package pkgid="2e9d4d23b7934b5d605e685969029fc0d8c7306ac4a9aebcaed97c8352d24776" name="gearmand" arch="src">
  <version epoch="0" ver="1.1.21" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.20-1" date="1675771200">- Update to release 1.1.20</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.21-1" date="1699358400">- Update to release 1.1.21
- Updated actions/checkout to v3 in GitHub Actions CI workflow
- Silence implicit fallthrough warning on newer GCC versions
- Added CXXFLAGS="-Wp,-D_GLIBCXX_ASSERTIONS" to gcc-9 and higher builds in GitHub
-   Actions CI
- More refinements to GitHub Actions CI
- Adjust linking order of libmemcached{,util}
- Change CI to test Ubuntu 18.04 in containers
- Add gcc-12 build to GitHub Actions CI
- skip removing entries from sqlite if store-queue-on-shutdown is used
- add Rocky to supported Linux distributions
- add loongarch64 support
- Issue #376: Fix configure error when gperf is absent
- Add riscv64 support
- added apt-get for some reqs in ubuntu</changelog>
</package>
<package pkgid="76e031a19b9e1bd94cd6368591d3f273d6be8129db737ae8a4321661401b2331" name="gflags" arch="src">
  <version epoch="0" ver="2.2.2" rel="16.el10"/>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.2.2-11" date="1674129600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_38_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.2.2-12" date="1689768000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_39_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.2.2-13" date="1705665600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_40_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.2.2-14" date="1706097600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_40_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.2.2-15" date="1721304000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_41_Mass_Rebuild</changelog>
  <changelog author="Miroslav Such&#xFD; &lt;msuchy@redhat.com&gt; - 2.2.2-16" date="1724846400">- convert license to SPDX</changelog>
</package>
<package pkgid="5f9964dd19550b4c1d74505e7f07adf64c49b1fbf0577196e7bc3ffe4d961211" name="gmime" arch="src">
  <version epoch="0" ver="2.6.23" rel="1.el10"/>
  <changelog author="Peter Robinson &lt;pbrobinson@fedoraproject.org&gt; 2.6.20-5" date="1431950400">- Rebuild (mono4)</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.6.20-5" date="1434542400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.6.20-6" date="1454500800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild</changelog>
  <changelog author="Kalev Lember &lt;klember@redhat.com&gt; - 2.6.20-7" date="1474545600">- BR vala instead of obsolete vala-tools subpackage</changelog>
  <changelog author="Kalev Lember &lt;klember@redhat.com&gt; - 2.6.22-1" date="1481976000">- Update to 2.6.22
- Use license macro for COPYING
- Tighten subpackage dependencies</changelog>
  <changelog author="Kalev Lember &lt;klember@redhat.com&gt; - 2.6.23-1" date="1485000000">- Update to 2.6.23</changelog>
</package>
<package pkgid="b90ed529bbaff3550b7fe13b6ac441ba651a8e2de12ce749b3edeaca7cda0e04" name="gperftools" arch="src">
  <version epoch="0" ver="2.6.1" rel="1.el10"/>
  <changelog author="Miroslav Rezanina &lt;mrezanin@redhat.com&gt; - 2.4-8.el7" date="1466596800">- gp-Use-initial-exec-tls-for-libunwind-s-recursion-flag.patch [bz#1339710]
- Resolves: bz#1339710
  (initalization of 'recursive' tls variable in libunwind stack capturer occasionally triggers deadlock in ceph)</changelog>
  <changelog author="Miroslav Rezanina &lt;mrezanin@redhat.com&gt; - 2.4-8.el7" date="1507809600">- Rebase to 2.6.1 [bz#1431240]
- Removed libunwind usage [bz#1467203]
- Resolves: bz#1431240 
  (gperftools fails to build on s390x, lacks s390x support)
- Resolves: bz#1467203
  (Please, remove libunwind from the gperftools-libs (and 389-ds-base) requirements)</changelog>
</package>
<package pkgid="a7c6a5df20c694813324bf7dd48ac13541043ad0d1ec53f40163d3508d9f64a7" name="gsl" arch="src">
  <version epoch="0" ver="2.7.1" rel="11.el10"/>
  <changelog author="Mamoru TASAKA &lt;mtasaka@fedoraproject.org&gt; - 2.7.1-2" date="1661256000">- Rebuild again for bodhi issue</changelog>
  <changelog author="Florian Weimer &lt;fweimer@redhat.com&gt; - 2.7.1-3" date="1674043200">- Port configure script to C99</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.7.1-4" date="1674129600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_38_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.7.1-5" date="1689854400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_39_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.7.1-6" date="1705752000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_40_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.7.1-7" date="1706097600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_40_Mass_Rebuild</changelog>
  <changelog author="David Abdurachmanov &lt;davidlt@rivosinc.com&gt; - 2.7.1-8" date="1708948800">- Disable FMA on riscv64</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.7.1-9" date="1721304000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_41_Mass_Rebuild</changelog>
  <changelog author="Miroslav Such&#xFD; &lt;msuchy@redhat.com&gt; - 2.7.1-10" date="1721908800">- convert license to SPDX</changelog>
  <changelog author="Jerry James &lt;loganjerry@gmail.com&gt; - 2.7.1-11" date="1723118400">- Verify the source tarball
- Stop converting THANKS to UTF-8; it already is
- Minor spec file cleanups</changelog>
</package>
<package pkgid="709b9dc913968ca551ad1150564b640a4beb1db6f98345ab35120e90f3e3f2fb" name="gtest" arch="src">
  <version epoch="0" ver="1.14.0" rel="5.el10"/>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.12.1-2" date="1658404800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_37_Mass_Rebuild</changelog>
  <changelog author="Terje Rosten &lt;terje.rosten@ntnu.no&gt; - 1.13.0-1" date="1674043200">- 1.13.0</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.13.0-2" date="1674129600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_38_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.13.0-3" date="1689854400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_39_Mass_Rebuild</changelog>
  <changelog author="Terje Rosten &lt;terje.rosten@ntnu.no&gt; - 1.14.0-1" date="1697371200">- 1.14.0</changelog>
  <changelog author="Benson Muite &lt;benson_muite@emailplus.org&gt; - 1.14.0-2" date="1700395200">- Use SPDX license identifiers</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.14.0-3" date="1705752000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_40_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.14.0-4" date="1706097600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_40_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.14.0-5" date="1721304000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_41_Mass_Rebuild</changelog>
</package>
<package pkgid="80cf121b457c0dae4daeadf627fcf9cad7d296a3572e824605e21bdb9c36838c" name="gtk2" arch="src">
  <version epoch="0" ver="2.24.33" rel="20.el10"/>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.24.33-11" date="1674129600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_38_Mass_Rebuild</changelog>
  <changelog author="Yaakov Selkowitz &lt;yselkowi@redhat.com&gt; - 2.24.33-12" date="1675252800">- Split out gtk-immodules-imsettings subpackage</changelog>
  <changelog author="Yaakov Selkowitz &lt;yselkowi@redhat.com&gt; - 2.24.33-13" date="1675252801">- Make the gtk-immodules-imsettings subpackage noarch</changelog>
  <changelog author="Florian Weimer &lt;fweimer@redhat.com&gt; - 2.24.33-14" date="1681214400">- Backport upstream patch to fix C99 compatibility issue</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.24.33-15" date="1689854400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_39_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.24.33-16" date="1705752000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_40_Mass_Rebuild</changelog>
  <changelog author="Florian Weimer &lt;fweimer@redhat.com&gt; - 2.24.33-17" date="1705838400">- Fix C89 compatibility issues, mostly via backports</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.24.33-18" date="1706097600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_40_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.24.33-19" date="1721304000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_41_Mass_Rebuild</changelog>
  <changelog author="Miroslav Such&#xFD; &lt;msuchy@redhat.com&gt; - 2.24.33-20" date="1725278400">- convert license to SPDX</changelog>
</package>
<package pkgid="b6af7fb0309ffe4e6e5c158fb8b8ecb4fae69c102d2e107c865b351c875ec9af" name="gtts-token" arch="src">
  <version epoch="0" ver="1.1.4" rel="8.el10"/>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.1.4-2" date="1611662400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_34_Mass_Rebuild</changelog>
  <changelog author="Python Maint &lt;python-maint@redhat.com&gt; - 1.1.4-3" date="1622808000">- Rebuilt for Python 3.10</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.1.4-4" date="1626955200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_35_Mass_Rebuild</changelog>
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- Add a fix for CVE-2017-11550 (#1478934)
- Add a fix for CVE-2004-2779 (#1561983)
- Use %license, remove Group tag
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.15.1b-29" date="1531483200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.15.1b-30" date="1549022400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.1.4-2" date="1689854400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_39_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.1.4-3" date="1705838400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_40_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.1.4-4" date="1706184000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_40_Mass_Rebuild</changelog>
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  <changelog author="Benjamin A. Beasley &lt;code@musicinmybrain.net&gt; 1.6.0-19" date="1626782400">- Drop EPEL workarounds</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; 1.6.0-20" date="1626955200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_35_Mass_Rebuild</changelog>
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  <changelog author="Benjamin A. Beasley &lt;code@musicinmybrain.net&gt; 1.6.0-23" date="1636632002">- Re-wrap description</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; 1.6.0-24" date="1642680000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_36_Mass_Rebuild</changelog>
  <changelog author="Benjamin A. Beasley &lt;code@musicinmybrain.net&gt; 1.6.0-25" date="1650110400">- Stop numbering patches</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; 1.6.0-26" date="1658404800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_37_Mass_Rebuild</changelog>
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  <changelog author="Benjamin A. Beasley &lt;code@musicinmybrain.net&gt; 1.6.0-21" date="1636632000">- Reduce macro indirection in the spec file</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; 1.6.0-26" date="1658404800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_37_Mass_Rebuild</changelog>
  <changelog author="Benjamin A. Beasley &lt;code@musicinmybrain.net&gt; 1.6.0-27" date="1663848000">- Correct License from GPLv2+ to GPLv2 with exceptions</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.6.1-1" date="1692187200">- Upgrade to version 1.6.1</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.1.3-26" date="1518004800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 0.1.3-27" date="1518696000">- Use ldconfig_scriptlets macro</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 0.1.3-28" date="1519041600">- Add missing BuildRequires on gcc and gcc-c++</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.1.3-29" date="1531483200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.1.3-30" date="1549022400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.1.3-31" date="1564056000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 0.1.3-32" date="1570622400">- Fix to build from source again</changelog>
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  <changelog author="Mohan Boddu &lt;mboddu@redhat.com&gt; - 2.3.0-6" date="1618574400">- Rebuilt for RHEL 9 BETA on Apr 15th 2021. Related: rhbz#1947937</changelog>
  <changelog author="Mohan Boddu &lt;mboddu@redhat.com&gt; - 2.3.0-7" date="1628510400">- Rebuilt for IMA sigs, glibc 2.34, aarch64 flags
  Related: rhbz#1991688</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.0-1" date="1675771200">- update to 2.4.0</changelog>
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  <changelog author="Mohan Boddu &lt;mboddu@redhat.com&gt; - 2.3.0-6" date="1618574400">- Rebuilt for RHEL 9 BETA on Apr 15th 2021. Related: rhbz#1947937</changelog>
  <changelog author="Mohan Boddu &lt;mboddu@redhat.com&gt; - 2.3.0-7" date="1628510400">- Rebuilt for IMA sigs, glibc 2.34, aarch64 flags
  Related: rhbz#1991688</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.0-1" date="1675771200">- update to 2.4.0</changelog>
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  <changelog author="Mohan Boddu &lt;mboddu@redhat.com&gt; - 2.3.0-6" date="1618574400">- Rebuilt for RHEL 9 BETA on Apr 15th 2021. Related: rhbz#1947937</changelog>
  <changelog author="Mohan Boddu &lt;mboddu@redhat.com&gt; - 2.3.0-7" date="1628510400">- Rebuilt for IMA sigs, glibc 2.34, aarch64 flags
  Related: rhbz#1991688</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.0-1" date="1675771200">- update to 2.4.0</changelog>
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  <changelog author="Mohan Boddu &lt;mboddu@redhat.com&gt; - 2.3.0-6" date="1618574400">- Rebuilt for RHEL 9 BETA on Apr 15th 2021. Related: rhbz#1947937</changelog>
  <changelog author="Mohan Boddu &lt;mboddu@redhat.com&gt; - 2.3.0-7" date="1628510400">- Rebuilt for IMA sigs, glibc 2.34, aarch64 flags
  Related: rhbz#1991688</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.5.0-1" date="1675771200">- update to 2.5.0</changelog>
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  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.5.0-1" date="1675771200">- update to 2.5.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.6.0-1" date="1708776000">- update to 2.6.0</changelog>
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  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 2.0.0-1" date="1570708800">- Update to upstream 2.0.0 release</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.0.1-1" date="1630065600">- Update to upstream 2.0.1 release</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.10.0-7" date="1674129600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_38_Mass_Rebuild</changelog>
  <changelog author="Paul Howarth &lt;paul@city-fan.org&gt; - 1.11.0-1" date="1685620800">- Update to 1.11.0 (rhbz#2211200)
  - Adds support for encrypt-then-mac (ETM) MACs
  - Adds support for AES-GCM crypto protocols
  - Adds support for sk-ecdsa-sha2-nistp256 and sk-ssh-ed25519 keys
  - Adds support for RSA certificate authentication
  - Adds FIDO support with *_sk() functions
  - Adds RSA-SHA2 key upgrading to OpenSSL, WinCNG, mbedTLS, OS400 backends
  - Adds Agent Forwarding and libssh2_agent_sign()
  - Adds support for Channel Signal message libssh2_channel_signal_ex()
  - Adds support to get the user auth banner message libssh2_userauth_banner()
  - Adds LIBSSH2_NO_{MD5, HMAC_RIPEMD, DSA, RSA, RSA_SHA1, ECDSA, ED25519,
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  - Adds direct stream UNIX sockets with libssh2_channel_direct_streamlocal_ex()
  - Adds wolfSSL support to CMake file
  - Adds mbedTLS 3.x support
  - Adds LibreSSL 3.5 support
  - Adds support for CMake "unity" builds
  - Adds CMake support for building shared and static libs in a single pass
  - Adds symbol hiding support to CMake
  - Adds support for libssh2.rc for all build tools
  - Adds .zip, .tar.xz and .tar.bz2 release tarballs
  - Enables ed25519 key support for LibreSSL 3.7.0 or higher
  - Improves OpenSSL 1.1 and 3 compatibility
  - Now requires OpenSSL 1.0.2 or newer
  - Now requires CMake 3.1 or newer
  - SFTP: Adds libssh2_sftp_open_ex_r() and libssh2_sftp_open_r() extended APIs
  - SFTP: No longer has a packet limit when reading a directory
  - SFTP: Now parses attribute extensions if they exist
  - SFTP: No longer will busy loop if SFTP fails to initialize
  - SFTP: Now clear various errors as expected
  - SFTP: No longer skips files if the line buffer is too small
  - SCP: Add option to not quote paths
  - SCP: Enables 64-bit offset support unconditionally
  - Now skips leading \r and \n characters in banner_receive()
  - Enables secure memory zeroing with all build tools on all platforms
  - No longer logs SSH_MSG_REQUEST_FAILURE packets from keepalive
  - Speed up base64 encoding by 7x
  - Assert if there is an attempt to write a value that is too large
  - WinCNG: fix memory leak in _libssh2_dh_secret()
  - Added protection against possible null pointer dereferences
  - Agent now handles overly large comment lengths
  - Now ensure KEX replies don't include extra bytes
  - Fixed possible buffer overflow when receiving SSH_MSG_USERAUTH_BANNER
  - Fixed possible buffer overflow in keyboard interactive code path
  - Fixed overlapping memcpy()
  - Fixed Windows UWP builds
  - Fixed DLL import name
  - Renamed local RANDOM_PADDING macro to avoid unexpected define on Windows
  - Support for building with gcc versions older than 8
  - Improvements to CMake, Makefile, NMakefile, GNUmakefile, autoreconf files
  - Restores ANSI C89 compliance
  - Enabled new compiler warnings and fixed/silenced them
  - Improved error messages
  - Now uses CIFuzz
  - Numerous minor code improvements
  - Improvements to CI builds
  - Improvements to unit tests
  - Improvements to doc files
  - Improvements to example files
  - Removed "old gex" build option
  - Removed no-encryption/no-mac builds
  - Removed support for NetWare and Watcom wmake build files
- Avoid use of deprecated patch syntax
- Build static library but don't package it since it's required for the
  test suite (https://github.com/libssh2/libssh2/issues/1056)
- Remove redundant references to %{_libdir} from pkgconfig file
- Add patch to work around strict permissions issues for sshd tests</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.11.0-2" date="1689854400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_39_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.11.0-3" date="1705838400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_40_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.11.0-4" date="1706184000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_40_Mass_Rebuild</changelog>
  <changelog author="Zhao Jiasheng &lt;JasenChao@gmail.com&gt; - 1.11.0-5" date="1712059200">- Fix rpath on riscv64</changelog>
  <changelog author="Paul Howarth &lt;paul@city-fan.org&gt; - 1.11.0-6" date="1720612800">- Build without OpenSSL ENGINE support from Fedora 41 onwards</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.11.0-7" date="1721304000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_41_Mass_Rebuild</changelog>
  <changelog author="Paul Howarth &lt;paul@city-fan.org&gt; - 1.11.0-8" date="1722081600">- Fix test suite failures with OpenSSH 9.8p1</changelog>
  <changelog author="Paul Howarth &lt;paul@city-fan.org&gt; - 1.11.1-1" date="1729080000">- Update to 1.11.1 (rhbz#2319104)
  - This is an enhancement and bugfix release - see RELEASE_NOTES for details
  - Note also that various algorithms are now deprecated and not built by
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</package>
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  <changelog author="Kate Hsuan &lt;hpa@redhat.com&gt; - 0.1.8-1" date="1668168000">- Update to 0.1.8-rc1</changelog>
  <changelog author="Kate Hsuan &lt;hpa@redhat.com&gt; - 0.1.8-1" date="1670587200">- update to 0.1.8</changelog>
  <changelog author="Kate Hsuan &lt;hpa@redhat.com&gt; - 0.1.8-3" date="1670846400">- Fix release ID.</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.1.8-4" date="1674129600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_38_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.1.8-5" date="1689854400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_39_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.1.8-6" date="1705838400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_40_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.1.8-7" date="1706184000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_40_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.1.8-8" date="1721304000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_41_Mass_Rebuild</changelog>
  <changelog author="Miroslav Such&#xFD; &lt;msuchy@redhat.com&gt; - 0.1.8-9" date="1725278400">- convert license to SPDX</changelog>
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  <changelog author="Wei-Lun Chao &lt;bluebat@member.fsf.org&gt; - 3.3" date="1665921600">- Rebuilt for Fedora</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.32.2-5" date="1658491200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_37_Mass_Rebuild</changelog>
  <changelog author="Joe Orton &lt;jorton@redhat.com&gt; - 0.32.3-1" date="1662379200">- update to 0.32.3</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.32.3-2" date="1674129600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_38_Mass_Rebuild</changelog>
  <changelog author="Joe Orton &lt;jorton@redhat.com&gt; - 0.32.5-1" date="1674475200">- update to 0.32.5</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.32.5-2" date="1689854400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_39_Mass_Rebuild</changelog>
  <changelog author="Joe Orton &lt;jorton@redhat.com&gt; - 0.32.5-3" date="1695988800">- migrated to SPDX license</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.32.5-4" date="1705838400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_40_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.32.5-5" date="1706184000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_40_Mass_Rebuild</changelog>
  <changelog author="Joe Orton &lt;jorton@redhat.com&gt; - 0.33.0-1" date="1706616000">- update to 0.33.0</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.33.0-2" date="1721304000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_41_Mass_Rebuild</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.1-9" date="1642680000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_36_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.1-10" date="1658491200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_37_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.1-11" date="1674129600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_38_Mass_Rebuild</changelog>
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  <changelog author="Remi Collet &lt;remi@remirepo.net&gt; - 3.2.14-2" date="1696420800">- build out of sources tree</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 3.2.15-2" date="1705838400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_40_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 3.2.15-3" date="1706184000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_40_Mass_Rebuild</changelog>
  <changelog author="Remi Collet &lt;remi@remirepo.net&gt; - 3.2.15-4" date="1713268800">- drop 32-bit support
  https://fedoraproject.org/wiki/Changes/php_no_32_bit</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 3.2.15-5" date="1721390400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_41_Mass_Rebuild</changelog>
  <changelog author="Remi Collet &lt;remi@remirepo.net&gt; - 3.2.16-1" date="1723464000">- update to 3.2.16 (no change)</changelog>
  <changelog author="Remi Collet &lt;remi@fedoraproject.org&gt; - 3.2.16-2" date="1728907200">- rebuild for https://fedoraproject.org/wiki/Changes/php84</changelog>
  <changelog author="Remi Collet &lt;remi@fedoraproject.org&gt; - 3.2.16-3" date="1728993600">- modernize spec file</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 3.0.0-8" date="1689940800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_39_Mass_Rebuild</changelog>
  <changelog author="Remi Collet &lt;remi@remirepo.net&gt; - 3.0.0-9" date="1696334400">- rebuild for https://fedoraproject.org/wiki/Changes/php83</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 3.0.0-10" date="1705838400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_40_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 3.0.0-11" date="1706184000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_40_Mass_Rebuild</changelog>
  <changelog author="Remi Collet &lt;remi@remirepo.net&gt; - 3.0.0-12" date="1713268800">- drop 32-bit support
  https://fedoraproject.org/wiki/Changes/php_no_32_bit</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 3.0.0-13" date="1721390400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_41_Mass_Rebuild</changelog>
  <changelog author="Remi Collet &lt;remi@fedoraproject.org&gt; - 3.0.0-14" date="1728907200">- rebuild for https://fedoraproject.org/wiki/Changes/php84</changelog>
  <changelog author="Remi Collet &lt;remi@fedoraproject.org&gt; - 3.0.0-15" date="1729684800">- modernize the spec file</changelog>
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  <changelog author="Remi Collet &lt;remi@remirepo.net&gt; - 3.2.0-5" date="1681992000">- use SPDX license ID</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 3.2.0-6" date="1689940800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_39_Mass_Rebuild</changelog>
  <changelog author="Remi Collet &lt;remi@remirepo.net&gt; - 3.2.0-7" date="1696334400">- rebuild for https://fedoraproject.org/wiki/Changes/php83</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 3.2.0-8" date="1705838400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_40_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 3.2.0-9" date="1706184000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_40_Mass_Rebuild</changelog>
  <changelog author="Remi Collet &lt;remi@remirepo.net&gt; - 3.2.0-9" date="1706616000">- build out of sources tree
- fix incompatible pointer types using patch from
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  <changelog author="Remi Collet &lt;remi@remirepo.net&gt; - 3.2.0-10" date="1713268800">- drop 32-bit support
  https://fedoraproject.org/wiki/Changes/php_no_32_bit</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 3.2.0-11" date="1721390400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_41_Mass_Rebuild</changelog>
  <changelog author="Remi Collet &lt;remi@fedoraproject.org&gt; - 3.2.0-13" date="1729080000">- modernize the spec file</changelog>
  <changelog author="Remi Collet &lt;remi@remirepo.net&gt; - 3.3.0-1" date="1729252800">- update to 3.3.0
- enable zstd compression support</changelog>
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  <changelog author="Remi Collet &lt;remi@remirepo.net&gt; - 2.2.0~RC2-2" date="1681992000">- use SPDX license ID</changelog>
  <changelog author="Remi Collet &lt;remi@remirepo.net&gt; - 2.2.0-1" date="1685707200">- update to 2.2.0
- build out of sources tree
- use parallel execution for test suite</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.2.0-2" date="1689940800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_39_Mass_Rebuild</changelog>
  <changelog author="Remi Collet &lt;remi@remirepo.net&gt; - 2.2.0-3" date="1696334400">- rebuild for https://fedoraproject.org/wiki/Changes/php83</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.2.0-4" date="1705838400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_40_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.2.0-5" date="1706184000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_40_Mass_Rebuild</changelog>
  <changelog author="Remi Collet &lt;remi@remirepo.net&gt; - 2.2.0-6" date="1713268800">- drop 32-bit support
  https://fedoraproject.org/wiki/Changes/php_no_32_bit</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.2.0-7" date="1721390400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_41_Mass_Rebuild</changelog>
  <changelog author="Remi Collet &lt;remi@remirepo.net&gt; - 3.0.0-1" date="1727352000">- update to 3.0.0
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 19-40" date="1674216000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_38_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 19-41" date="1689940800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_39_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 19-42" date="1705838400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_40_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 19-43" date="1706184000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_40_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 19-44" date="1721390400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_41_Mass_Rebuild</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.8-3" date="1501761600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Binutils_Mass_Rebuild</changelog>
  <changelog author="Nikos Mavrogiannopoulos &lt;nmav@redhat.com&gt; - 1.2.9-1" date="1516622400">- New upstream release</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.9-2" date="1518177600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Nikos Mavrogiannopoulos &lt;nmav@redhat.com&gt; - 1.2.10-1" date="1526385600">- New upstream release</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.10-2" date="1531569600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Nikos Mavrogiannopoulos &lt;nmav@redhat.com&gt; - 1.2.11-1" date="1536926400">- New upstream release</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.11-2" date="1549108800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.11-3" date="1564142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
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  <changelog author="Nikos Mavrogiannopoulos &lt;nmav@redhat.com&gt; - 1.2.11-1" date="1536926400">- New upstream release</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.11-2" date="1549108800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.11-3" date="1564142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.12-1" date="1599825600">- New upstream release</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.11-3" date="1564142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.12-1" date="1599825600">- New upstream release</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.1.1-6" date="1658577600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_37_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.1.1-7" date="1674216000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_38_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.1.1-8" date="1689940800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_39_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.1.1-9" date="1706270400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_40_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.1.1-10" date="1721390400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_41_Mass_Rebuild</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 13.0.3-200.1" date="1611748800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_34_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 13.0.3-200.2" date="1627041600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_35_Mass_Rebuild</changelog>
  <changelog author="Mamoru TASAKA &lt;mtasaka@fedoraproject.org&gt; - 13.0.6-200" date="1627560000">- 13.0.6</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 13.0.6-200.1" date="1642766400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_36_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 13.0.6-200.2" date="1658577600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_37_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 13.0.6-200.3" date="1674216000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_38_Mass_Rebuild</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.2.0.2-3" date="1611748800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_34_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.2.0.2-4" date="1627041600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_35_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.2.0.2-5" date="1642766400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_36_Mass_Rebuild</changelog>
  <changelog author="V&#xED;t Ondruch &lt;vondruch@redhat.com&gt; - 2.2.0.2-6" date="1643198400">- Rebuilt for https://fedoraproject.org/wiki/Changes/Ruby_3.1</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.2.0.2-7" date="1658577600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_37_Mass_Rebuild</changelog>
  <changelog author="Mamoru TASAKA &lt;mtasaka@fedoraproject.org&gt; - 2.2.0.2-8" date="1671969600">- Use %gem_extdir_mri instead of ext for %check due to ruby3.2 change
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.2.0.2-10" date="1674216000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_38_Mass_Rebuild</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.7.3-10" date="1501156800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.7.3-11" date="1518177600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.7.3-12" date="1531569600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.7.3-13" date="1549108800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
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  <changelog author="Mamoru TASAKA &lt;mtasaka@fedoraproject.org&gt; - 4.1.0-1" date="1676635200">- 4.1.0</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 4.1.0-2" date="1689940800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_39_Mass_Rebuild</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 4.2.0-2" date="1706270400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_40_Mass_Rebuild</changelog>
  <changelog author="Mamoru TASAKA &lt;mtasaka@fedoraproject.org&gt; - 4.2.1-1" date="1710763200">- 4.2.1</changelog>
  <changelog author="Mamoru TASAKA &lt;mtasaka@fedoraproject.org&gt; - 4.3.0-1" date="1718712000">- 4.3.0</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 4.3.0-2" date="1721390400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_41_Mass_Rebuild</changelog>
  <changelog author="Mamoru TASAKA &lt;mtasaka@fedoraproject.org&gt; - 4.4.0-1" date="1726833600">- 4.4.0</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 3.10.0-2" date="1611748800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_34_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 3.10.0-3" date="1627041600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_35_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 3.10.0-4" date="1642766400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_36_Mass_Rebuild</changelog>
  <changelog author="Mamoru TASAKA &lt;mtasaka@fedoraproject.org&gt; - 3.11.0-1" date="1644494400">- 3.11.0</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 3.11.0-2" date="1658577600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_37_Mass_Rebuild</changelog>
  <changelog author="Mamoru TASAKA &lt;mtasaka@fedoraproject.org&gt; - 3.12.0-1" date="1666872000">- 3.12.0</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 3.12.0-2" date="1674216000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_38_Mass_Rebuild</changelog>
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  <changelog author="V&#xED;t Ondruch &lt;vondruch@redhat.com&gt; - 1.4.2-8" date="1643198400">- Rebuilt for https://fedoraproject.org/wiki/Changes/Ruby_3.1</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.4.2-10" date="1658577600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_37_Mass_Rebuild</changelog>
  <changelog author="V&#xED;t Ondruch &lt;vondruch@redhat.com&gt; - 1.4.2-11" date="1672747200">- Rebuilt for https://fedoraproject.org/wiki/Changes/Ruby_3.2</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.4.2-12" date="1674216000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_38_Mass_Rebuild</changelog>
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  <changelog author="Robert Scheck &lt;robert@fedoraproject.org&gt; 0.2.0-18" date="1659268800">- Added sysusers.d file to achieve user() and group() provides</changelog>
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  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.7-1" date="1605009600">- Updated to release 1.4.7</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.8-1" date="1605009601">- Updated to release 1.4.8</changelog>
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  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.8-1" date="1605009601">- Updated to release 1.4.8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.9-1" date="1621512000">- Updated to release 1.4.9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.10-1" date="1637582400">- Updated to release 1.4.10</changelog>
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  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.6.0-1" date="1661947200">- Updated to release 1.6.0
- capture: added compatibility with openssl &gt;= 3.5.0
- capture: fixed memory leak while rotating dialogs
- capture: added save file rotation through SIPHUP signal
- capture: added support for opening gzip compressed input files
- hep: received HEP packets can now be saved to PCAP files
- core: added support for PCRE2 library
- cf: raw payload preview now properly displays characters near window borders
- cli: properly display captured dialog count when no interface is used</changelog>
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  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.9-1" date="1621512000">- Updated to release 1.4.9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.10-1" date="1637582400">- Updated to release 1.4.10</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.5.0-1" date="1650974400">- Updated to release 1.5.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.6.0-1" date="1661947200">- Updated to release 1.6.0
- capture: added compatibility with openssl &gt;= 3.5.0
- capture: fixed memory leak while rotating dialogs
- capture: added save file rotation through SIPHUP signal
- capture: added support for opening gzip compressed input files
- hep: received HEP packets can now be saved to PCAP files
- core: added support for PCRE2 library
- cf: raw payload preview now properly displays characters near window borders
- cli: properly display captured dialog count when no interface is used</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.7.0-1" date="1680264000">- Updated to release 1.7.0
- save: add option --text to save captured data to plain text
- capture: fix memory overflows while parsing IP headers
- hep: fix hep listener enabled in offline mode
- core: stop sngrep when parent process has ended
- ssl: fix decrypt with AES256 GCM SHA384 cipher</changelog>
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  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.6.0-1" date="1661947200">- Updated to release 1.6.0
- capture: added compatibility with openssl &gt;= 3.5.0
- capture: fixed memory leak while rotating dialogs
- capture: added save file rotation through SIPHUP signal
- capture: added support for opening gzip compressed input files
- hep: received HEP packets can now be saved to PCAP files
- core: added support for PCRE2 library
- cf: raw payload preview now properly displays characters near window borders
- cli: properly display captured dialog count when no interface is used</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.7.0-1" date="1680264000">- Updated to release 1.7.0
- save: add option --text to save captured data to plain text
- capture: fix memory overflows while parsing IP headers
- hep: fix hep listener enabled in offline mode
- core: stop sngrep when parent process has ended
- ssl: fix decrypt with AES256 GCM SHA384 cipher</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.8.0-1" date="1703073600">- Updated to release 1.8.0
- fix typo in message, thanks to lintian
- fix compiler warnings about unused variables
- [fix] debian pkg dependency to libpcre2-8-0
- Fixed a typo in comment line in filter.c
- Redefine usage of POSIX signals.
- Support for building sngrep using CMake</changelog>
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- capture: added compatibility with openssl &gt;= 3.5.0
- capture: fixed memory leak while rotating dialogs
- capture: added save file rotation through SIPHUP signal
- capture: added support for opening gzip compressed input files
- hep: received HEP packets can now be saved to PCAP files
- core: added support for PCRE2 library
- cf: raw payload preview now properly displays characters near window borders
- cli: properly display captured dialog count when no interface is used</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.7.0-1" date="1680264000">- Updated to release 1.7.0
- save: add option --text to save captured data to plain text
- capture: fix memory overflows while parsing IP headers
- hep: fix hep listener enabled in offline mode
- core: stop sngrep when parent process has ended
- ssl: fix decrypt with AES256 GCM SHA384 cipher</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.8.0-1" date="1703073600">- Updated to release 1.8.0
- fix typo in message, thanks to lintian
- fix compiler warnings about unused variables
- [fix] debian pkg dependency to libpcre2-8-0
- Fixed a typo in comment line in filter.c
- Redefine usage of POSIX signals.
- Support for building sngrep using CMake</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.8.1-1" date="1712577600">- Updated to release 1.8.1
- Fixes for Stack Buffer Overflows in SIP Header Processing</changelog>
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  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.6.0-1" date="1661947200">- Updated to release 1.6.0
- capture: added compatibility with openssl &gt;= 3.5.0
- capture: fixed memory leak while rotating dialogs
- capture: added save file rotation through SIPHUP signal
- capture: added support for opening gzip compressed input files
- hep: received HEP packets can now be saved to PCAP files
- core: added support for PCRE2 library
- cf: raw payload preview now properly displays characters near window borders
- cli: properly display captured dialog count when no interface is used</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.7.0-1" date="1680264000">- Updated to release 1.7.0
- save: add option --text to save captured data to plain text
- capture: fix memory overflows while parsing IP headers
- hep: fix hep listener enabled in offline mode
- core: stop sngrep when parent process has ended
- ssl: fix decrypt with AES256 GCM SHA384 cipher</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.8.0-1" date="1703073600">- Updated to release 1.8.0
- fix typo in message, thanks to lintian
- fix compiler warnings about unused variables
- [fix] debian pkg dependency to libpcre2-8-0
- Fixed a typo in comment line in filter.c
- Redefine usage of POSIX signals.
- Support for building sngrep using CMake</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.8.1-1" date="1712577600">- Updated to release 1.8.1
- Fixes for Stack Buffer Overflows in SIP Header Processing</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.8.2-1" date="1720440000">- Updated to release 1.8.2
- Fixed heap buffer overflow while parsing RTP packets payload</changelog>
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  <changelog author="Jiri Kucera &lt;jkucera@redhat.com&gt; - 14.4.2.0-20" date="1521633600">- added patch that fixes WAV to HCOM conversion abortion on 64 bit big endian
  machines
  + resolves #1558887</changelog>
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- prevents division by zero in src/ao.c
  + fixes/prevents "sox killed by SIGFPE (signal 8)" kind of bugs that appear
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  + related bugs: #1309426, #1226675, #1540762, #1492910</changelog>
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  + "divide by zero in startread function in wav.c" (CVE-2017-11332)
  + "invalid memory read in read_samples function in hcom.c" (CVE-2017-11358)
  + "divide by zero in wavwritehdr function in wav.c" (CVE-2017-11359)
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 14.4.2.0-24" date="1531569600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 14.4.2.0-25" date="1549195200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 14.4.2.0-26" date="1564142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.5-7" date="1549195200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.5-8" date="1564142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.5-12" date="1580472000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_32_Mass_Rebuild</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.3.0-10" date="1690027200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_39_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.3.0-11" date="1706356800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_40_Mass_Rebuild</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.3.0-4" date="1658577600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_37_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.3.0-5" date="1674302400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_38_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.3.0-6" date="1690027200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_39_Mass_Rebuild</changelog>
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  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 3.0.0-1" date="1720612800">- Update to 3.0.0</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 3.0.0-2" date="1721476800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_41_Mass_Rebuild</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.51.08-3" date="1690027200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_39_Mass_Rebuild</changelog>
  <changelog author="Jonathan Wright &lt;jonathan@almalinux.org&gt; - 1.59.02-1" date="1704283200">- Update to 1.59.02 rhbz#1694044</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.59.02-2" date="1706356800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_40_Mass_Rebuild</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.59.03-2" date="1721476800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_41_Mass_Rebuild</changelog>
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- fixes rhbz#2310136
- fixes rhbz#2310146
- fixes rhbz#2310152</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.8.1-6" date="1690027200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_39_Mass_Rebuild</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.8.2-3" date="1721476800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_41_Mass_Rebuild</changelog>
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