Log message:
Update to Asterisk 23.3.0:

## 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)

### Summary:

- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0

### 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

- #### res_rtp_asterisk.c: Fix DTLS packet drop when TURN loopback re-injection \ 
occurs before ICE candidate check
  WebRTC calls using TURN configured in rtp.conf (turnaddr,
  turnusername, turnpassword) will now correctly complete DTLS/SRTP
  negotiation. Previously all DTLS packets were silently dropped due to
  the loopback re-injection address not being in the ICE active candidate
  list.

- #### docs: Add "Provided-by" to doc XML and CLI output.
  The CLI help for applications, functions, manager commands and
  manager events now shows the module that provides its functionality.

- #### CDR/CEL Custom Performance Improvements
  Significant performance improvements have been made to the
  cdr_custom, cdr_sqlite3_custom, cel_custom and cel_sqlite3_custom modules.
  See the new sample config files for those modules to see how to benefit
  from them.

- #### chan_websocket: Add media direction.
  WebSocket now supports media direction, allowing for
  unidirectional media. This is done from the perspective of the
  application and can be set via channel origination, external media, or
  commands sent from the application. Check out
  https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket/ for
  more.

- #### app_queue: Add 'prio' setting to the 'force_longest_waiting_caller' option
  The 'force_longest_waiting_caller' option now supports a 'prio' setting.
  When set to 'prio', calls are offered by priority first, then by wait time.

- #### Upgrade bundled pjproject to 2.16.
  Bundled pjproject has been upgraded to 2.16. 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_header_funcs: Add new PJSIP_INHERITABLE_HEADER dialplan function
  A new PJSIP_HEADER option has been added that allows
  inheriting pjsip headers from the inbound to the outbound bridged
  channel.
  Example- same => n,Set(PJSIP_INHERITABLE_HEADER(add,X-custom-1)=alpha)
  will add X-custom-1: alpha to the outbound pjsip channel INVITE
  upon Dial.

- #### app_queue: Fix rN raise_penalty ignoring min_penalty in calc_metric
  Fixes an issue where QUEUE_RAISE_PENALTY=rN could raise a member’s penalty \ 
below QUEUE_MIN_PENALTY during member selection. This could allow members \ 
intended to be excluded to be selected. The queue now consistently respects the \ 
minimum penalty when raising penalties, aligning member selection behavior with \ 
queue empty checks and documented rN semantics.

## Issue and Commit Detail:

### Closed Issues:

  - 449: [bug]:  PJSIP confuses media address after INVITE requiring authentication
  - 566: [bug]: core: SIGSEGV on DTMF when no timing modules loaded
  - 1356: [bug]: MESSAGE requests should not contain a Contact header
  - 1524: [bug]: PJSIP if sdp_session is blank the initial INVITE doesn't attach \ 
an SDP offer, worked in chan_sip
  - 1611: [bug]: asterisk deadlocked on start sometimes
  - 1612: [improvement]: pjsip: Upgrade bundled version to pjproject 2.16
  - 1637: [improvement]: force_longest_waiting_caller should also consider \ 
caller priority
  - 1641: [bug]: res_pjsip_config_wizard: Endpoints fail to update when Named \ 
ACLs change after reload
  - 1651: [bug]: Asterisk crashes with munmap_chunk() when using sorcery \ 
realtime for PJSIP registration objects
  - 1657: [bug]: Wrong dtmf payload is used when inbound invite contains 8K and \ 
16K, and outgoing leg is using G722 and SRTP
  - 1670: [new-feature]: Add new option to PJSIP_HEADER to pass headers from the \ 
inbound to outbound channel.
  - 1691: [bug]: force_longest_waiting_caller stops offering calls if a call \ 
joins at the first position
  - 1703: [bug]: res_pjsip_pubsub: ao2 reference leak of subscription tree in \ 
ast_sip_subscription
  - 1707: [bug]: chan_iax2: Crash when processing video frames with negative length
  - 1716: [bug]: Ghost call when UAC didn't respond with 487 for a cancel \ 
request from server even after original call hangup.
  - 1724: [improvement]: say.c - added language support for pashto and dari
  - 1730: [bug]: CPP channel storage get_by_name_prefix does not check prefix match
  - 1755: [bug]: app_dial, utils.h: Compilation failure with \ 
-Wold-style-declaration and -Wdiscarded-qualifiers
  - 1781: [bug]: More discarded-qualifiers errors with gcc 15.2.1
  - 1783: [bug]: Several unused-but-set-variable warnings with gcc 16
  - 1785: [bug]: chan_websocket doesn’t work with genericplc and transcoding
  - 1786: [bug]: chan_dahdi: A few more discarded-qualifiers errors not caught \ 
previously
  - 1795: [bug]: DTLS packets dropped when TURN configured in rtp.conf due to \ 
loopback re-injection occurring before ICE candidate source check
  - 1797: [bug]: Potential logic issue in translated frame write loop (main/file.c)
  - 1802: [improvement]: app_dial: Channel name should be included in warnings \ 
during wait_for_answer
  - 1804: [new-feature]: dsp.c: Add support for R2 signaling
  - 1814: [bug]: A pjsip transport with an invalid config can cause issues with \ 
other transports
  - 1816: [bug]: ARI: RTPAUDIO channel vars aren't set if call hung up by ARI.
  - 1819: [bug]: When a 302 is received from a UAS, the cause and tech_cause \ 
codes set on the channel are incorrect.
  - 1831: [bug]:raise_exception() and EXCEPTION() read use channel datastores \ 
without holding ast_channel_lock
  - 1833: [bug]: Address security vulnerabilities in pjproject
  - 1844: [bug]: cdrel_custom isn't respecting the default time format for CEL \ 
records
  - 1845: [bug]:res_cdrel_custom produces wrong float timestamps
  - 1852: [bug]: res_cdrel_custom: connection to the sqlite3 database closes \ 
from time to time

### Commit List:

-  res_cdrel_custom: do not free config when no new config was loaded
-  res_cdrel_custom: Resolve several formatting issues.
-  res_pjsip: Address pjproject security vulnerabilities
-  pbx: Hold channel lock for exception datastore access
-  xmldoc.c: Fix memory leaks in handling of provided_by.
-  SECURITY.md: Update with additional instructions.
-  res_audiosocket: Fix header read loop to use correct buffer offset
-  manager.c : Fix CLI event display
-  chan_pjsip: Set correct cause codes for non-2XX responses.
-  res_pjsip_config_wizard: Force reload on Named ACL change events
-  rtp: Set RTPAUDIOQOS variables when ast_softhangup is called.
-  channel: Prevent crash during DTMF emulation when no timing module is loaded
-  res_pjsip: Remove temp transport state when a transport fails to load.
-  res_pjsip_messaging: Remove Contact header from out-of-dialog MESSAGE as per \ 
RFC3428
-  acl: Add ACL support to http and ari
-  res_rtp_asterisk.c: Fix DTLS packet drop when TURN loopback re-injection \ 
occurs before ICE candidate check
-  dsp.c: Add support for detecting R2 signaling tones.
-  app_dial: Include channel name in warnings during wait_for_answer.
-  main/file: fix translated-frame write loop to use current frame
-  docs: Add "Provided-by" to doc XML and CLI output.
-  chan_websocket_doc.xml: Add d(media_direction) option.
-  resource_channels.c: Fix validation response for externalMedia with AudioSockets
-  CDR/CEL Custom Performance Improvements
-  chan_websocket: Remove silence generation and frame padding.
-  chan_websocket: Add media direction.
-  fix: Add macOS (Darwin) compatibility for building Asterisk
-  astconfigparser.py: Fix regex pattern error by properly escaping string
-  res_rtp_asterisk: use correct sample rate lookup to account for g722
-  res_pjsip_outbound_registration.c: Prevent crash if load_module() fails
-  pjsip_configuration: Ensure s= and o= lines in SDP are never empty
-  res_pjsip_session: Make sure NAT hook runs when packet is retransmitted for \ 
whatever reason.
-  chan_dahdi: Fix discarded-qualifiers errors.
-  build: Fix unused-but-set-variable warnings with gcc 16.
-  build: Fix another GCC discarded-qualifiers const error.
-  chan_iax2: Fix crash due to negative length frame lengths.
-  build: Fix GCC discarded-qualifiers const errors.
-  endpoints: Allow access to latest snapshot directly.
-  app_dial, utils.h: Avoid old style declaration and discarded qualifier.
-  app_queue: Add 'prio' setting to the 'force_longest_waiting_caller' option
-  Upgrade bundled pjproject to 2.16.
-  res_pjsip_header_funcs: Add new PJSIP_INHERITABLE_HEADER dialplan function
-  app_queue: Queue Timing Parity with Dial() and Accurate Wait Metrics
-  stasis.c: Fix deadlock in stasis_topic_pool_get_topic during module load
-  app_queue: Fix rN raise_penalty ignoring min_penalty in calc_metric
-  app_queue: Only compare calls at 1st position across queues when forcing \ 
longest waiting caller.
-  channelstorage_cpp_map_name_id: Fix get_by_name_prefix prefix match
-  app_amd: Remove errant space in documentation for totalAnalysisTime.
-  say.c: added language support for pashto and dari
-  res_pjsip_session.c: Prevent INVITE failover when session is cancelled
-  res_pjsip_pubsub: Fix ao2 reference leak of subscription tree in \ 
ast_sip_subscription

Log message:
*: remove OWNER definition

OWNER, when it was introduced, was to protect packages deep in the
infrastructure by emphasizing that they should not be touched by
non-MAINTAINERs.

No infrastructure package still sets OWNER.

Note: non-trivial change to packages should be passed by MAINTAINERs.

As discussed on tech-pkg.

Log message:
update to Asterisk 23.2.2:

----- 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)

### 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.

----- 23.2.1 -----

## 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)

### Summary:

- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0

## 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

----- 23.2.0 -----

## 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)

### 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 `<deviceID>` 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 <taskprocessor-name>

- #### 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(<seconds>)' (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 to the

Log message:
*: recursive bump for nettle 4.0 shlib major bump

Log message:
*: recursive bump for icu 78.1

Log message:
Update to Asterisk 23.1.0.

## 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)

### 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 > 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.

## Issue and Commit Detail:

### Closed Issues:

  - 781: [improvement]: Allow call by call disabling Stir/Shaken header inclusion
  - 1340: [bug]: comfort noise packet corrupted
  - 1419: [bug]: static code analysis issues in app_adsiprog.c
  - 1422: [bug]: static code analysis issues in apps/app_externalivr.c
  - 1425: [bug]: static code analysis issues in apps/app_queue.c
  - 1434: [improvement]: pbx_variables: Create real channel for dialplan eval \ 
CLI command
  - 1436: [improvement]: res_cliexec: Avoid unnecessary cast to char*
  - 1451: [bug]: ast_config_text_file_save2(): incorrect handling of deep/wide \ 
template inheritance
  - 1455: [new-feature]: chan_dahdi: Add DAHDI_CHANNEL function
  - 1467: [bug]: Crash in res_pjsip_refer during REFER progress teardown with \ 
PJSIP_TRANSFER_HANDLING(ari-only)
  - 1478: [improvement]: Stasis threadpool -> taskpool
  - 1479: [bug]: The ARI bridge play and record APIs limit audio bandwidth by \ 
forcing the slin8 format.
  - 1483: [improvement]: sig_analog: Eliminate possible timeout for Last Number \ 
Redial
  - 1485: [improvement]: func_scramble: Add example to XML documentation.
  - 1487: [improvement]: app_dial: Allow partial seconds to be used for dial timeouts
  - 1489: [improvement]: config_options.c: Improve misleading error message
  - 1491: [bug]: Segfault: `channelstorage_cpp` fast lookup without lock \ 
(`get_by_name_exact`/`get_by_uniqueid`) leads to UAF during hangup
  - 1493: [new-feature]: app_sf: Add post-digit timer option
  - 1496: [improvement]: dsp.c: Minor fixes to debug log messages
  - 1499: [new-feature]: func_math: Add function to return the digit sum
  - 1501: [improvement]: codec_builtin: Fix some inaccurate quality weights.
  - 1505: [improvement]: res_fax: Add XML documentation for channel variables
  - 1507: [improvement]: res_tonedetect: Minor formatting issue in documentation
  - 1509: [improvement]: res_fax.c — log debug error as debug, not regular log
  - 1510: [new-feature]: sig_analog: Allow '#' to end the inter-digit timeout \ 
when dialing.
  - 1514: [improvement]: func_channel: Allow R/W of ADSI CPE capability setting.
  - 1517: [improvement]: core_unreal: Preserve ADSI capability when dialing \ 
Local channels
  - 1519: [improvement]: app_dial / func_callerid: DNIS information is not \ 
propagated by Dial
  - 1525: [bug]: chan_websocket: fix use of raw payload variable for string \ 
comparison in process_text_message
  - 1534: [bug]: app_queue when using gosub breaks dialplan when going from 20 \ 
to 21, What's new in 21 doesn't mention it's a breaking change,
  - 1535: [bug]: chan_pjsip changes SSRC on WebRTC channels, which is \ 
unsupported by some browsers
  - 1536: [bug]: asterisk -rx connects to console instead of executing a command
  - 1539: [bug]: safe_asterisk without TTY doesn't log to file
  - 1544: [improvement]: While Receiving the MediaConnect Message Using External \ 
Media Over websocket ChannelID is  Details are missing
  - 1554: [bug]: safe_asterisk recurses into subdirectories of startup.d after f97361
  - 1559: [improvement]: Handle TLS handshake attacks in order to resolve the \ 
issue of exceeding the maximum number of HTTPS sessions.
  - 1578: [bug]: Deadlock with externalMedia custom channel id and cpp map \ 
channel backend

Log message:
comms/asterisk23: import asterisk-23.0.0

Asterisk is a complete PBX in software.  It provides all of the
features you would expect from a PBX and more. Asterisk does voice
over IP in three protocols, and can interoperate with almost all
standards-based telephony equipment using relatively inexpensive
hardware.

Asterisk provides Voicemail services with Directory, Call Conferencing,
Interactive Voice Response, Call Queuing. It has support for
three-way calling, caller ID services, ADSI, SIP and H.323 (as both
client and gateway).

This is a standard version.  It is scheduled to go to security
fixes only on October 15th, 2026, and EOL on October 15th, 2027.
See here for more information about Asterisk versions:
https://docs.asterisk.org/About-the-Project/Asterisk-Versions/