The Asterisk Development Team would like to announce
release candidate 1 of asterisk-23.2.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/23.2.0-rc1
and
https://downloads.asterisk.org/pub/telephony/asterisk

Repository: https://github.com/asterisk/asterisk
Tag: 23.2.0-rc1

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-rc1

Links:

• Full ChangeLog
• GitHub Diff
• Tarball
• Downloads

Summary:

• Commits: 57
• Commit Authors: 20
• Issues Resolved: 41
• Security Advisories Resolved: 0

User Notes:

• 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

• 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()' (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

Commit Authors:

• Alexei Gradinari: (1)
• C. Maj: (1)
• Daouda Taha: (1)
• George Joseph: (11)
• 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)

The Asterisk Development Team would like to announce
release candidate 1 of asterisk-22.8.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/22.8.0-rc1
and
https://downloads.asterisk.org/pub/telephony/asterisk

Repository: https://github.com/asterisk/asterisk
Tag: 22.8.0-rc1

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-rc1

Links:

• Full ChangeLog
• GitHub Diff
• Tarball
• Downloads

Summary:

• Commits: 56
• Commit Authors: 19
• Issues Resolved: 40
• Security Advisories Resolved: 0

User Notes:

• 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

• 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()' (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

Commit Authors:

• Alexei Gradinari: (1)
• C. Maj: (1)
• Daouda Taha: (1)
• George Joseph: (11)
• 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)

The Asterisk Development Team would like to announce
release candidate 1 of asterisk-20.18.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/20.18.0-rc1
and
https://downloads.asterisk.org/pub/telephony/asterisk

Repository: https://github.com/asterisk/asterisk
Tag: 20.18.0-rc1

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-rc1

Links:

• Full ChangeLog
• GitHub Diff
• Tarball
• Downloads

Summary:

• Commits: 56
• Commit Authors: 20
• Issues Resolved: 40
• Security Advisories Resolved: 0

User Notes:

• 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

• 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()' (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

Commit Authors:

• Alexei Gradinari: (1)
• C. Maj: (1)
• Daouda Taha: (1)
• Etienne Lessard: (1)
• George Joseph: (11)
• 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)

The Asterisk Development Team would like to announce
the release of Certified asterisk-20.7-cert8.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert8
and
https://downloads.asterisk.org/pub/telephony/certified-asterisk

Repository: https://github.com/asterisk/asterisk
Tag: certified-20.7-cert8

This release resolves issues reported by the community
and would have not been possible without your participation.

Thank You!

Change Log for Release asterisk-certified-20.7-cert8

Links:

• Full ChangeLog
• GitHub Diff
• Tarball
• Downloads

Summary:

• Commits: 7
• Commit Authors: 3
• Issues Resolved: 7
• Security Advisories Resolved: 0

User Notes:

• 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.

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})
  

Developer Notes:

Commit Authors:

• George Joseph: (4)
• Mike Bradeen: (2)
• Sean Bright: (1)

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
• GitHub Diff
• Tarball
• Downloads

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 > 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)

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
• GitHub Diff
• Tarball
• Downloads

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.

Developer Notes:

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)

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
• GitHub Diff
• Tarball
• Downloads

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.

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)

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
• GitHub Diff
• Tarball
• Downloads

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 > 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)

The Asterisk Development Team would like to announce
release candidate 2 of asterisk-23.1.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/23.1.0-rc2
and
https://downloads.asterisk.org/pub/telephony/asterisk

Repository: https://github.com/asterisk/asterisk
Tag: 23.1.0-rc2

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-rc2

Links:

• Full ChangeLog
• GitHub Diff
• Tarball
• Downloads

Summary:

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

User Notes:

Upgrade Notes:

Developer Notes:

Commit Authors:

• George Joseph: (1)

Issue and Commit Detail:

Closed Issues:

• 1578: [bug]: Deadlock with externalMedia custom channel id and cpp map channel backend

Commits By Author:

• George Joseph (1):

Commit List:

• channelstorage: Allow storage driver read locking to be skipped.

Commit Details:

channelstorage: Allow storage driver read locking to be skipped.

Author: George Joseph
Date: 2025-11-06

After PR #1498 added read locking to channelstorage_cpp_map_name_id, if ARI
channels/externalMedia was called with a custom channel id AND the
cpp_map_name_id channel storage backend is in use, a deadlock can occur when
hanging up the channel. It's actually triggered in
channel.c:__ast_channel_alloc_ap() when it gets a write lock on the
channelstorage driver then subsequently does a lookup for channel uniqueid
which now does a read lock. This is an invalid operation and causes the lock
state to get "bad". When the channels try to hang up, a write lock is
attempted again which hangs and causes the deadlock.

Now instead of the cpp_map_name_id channelstorage driver "get" APIs
automatically performing a read lock, they take a "lock" parameter which
allows a caller who already has a write lock to indicate that the "get" API
must not attempt its own lock. This prevents the state from getting mesed up.

The ao2_legacy driver uses the ao2 container's recursive mutex so doesn't
have this issue but since it also implements the common channelstorage API,
it needed its "get" implementations updated to take the lock parameter. They
just don't use it.

Resolves: #1578

The Asterisk Development Team would like to announce
release candidate 2 of asterisk-22.7.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/22.7.0-rc2
and
https://downloads.asterisk.org/pub/telephony/asterisk

Repository: https://github.com/asterisk/asterisk
Tag: 22.7.0-rc2

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-rc2

Links:

• Full ChangeLog
• GitHub Diff
• Tarball
• Downloads

Summary:

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

User Notes:

Upgrade Notes:

Developer Notes:

Commit Authors:

• George Joseph: (1)

Issue and Commit Detail:

Closed Issues:

• 1578: [bug]: Deadlock with externalMedia custom channel id and cpp map channel backend

Commits By Author:

• George Joseph (1):

Commit List:

• channelstorage: Allow storage driver read locking to be skipped.

Commit Details:

channelstorage: Allow storage driver read locking to be skipped.

Author: George Joseph
Date: 2025-11-06

After PR #1498 added read locking to channelstorage_cpp_map_name_id, if ARI
channels/externalMedia was called with a custom channel id AND the
cpp_map_name_id channel storage backend is in use, a deadlock can occur when
hanging up the channel. It's actually triggered in
channel.c:__ast_channel_alloc_ap() when it gets a write lock on the
channelstorage driver then subsequently does a lookup for channel uniqueid
which now does a read lock. This is an invalid operation and causes the lock
state to get "bad". When the channels try to hang up, a write lock is
attempted again which hangs and causes the deadlock.

Now instead of the cpp_map_name_id channelstorage driver "get" APIs
automatically performing a read lock, they take a "lock" parameter which
allows a caller who already has a write lock to indicate that the "get" API
must not attempt its own lock. This prevents the state from getting mesed up.

The ao2_legacy driver uses the ao2 container's recursive mutex so doesn't
have this issue but since it also implements the common channelstorage API,
it needed its "get" implementations updated to take the lock parameter. They
just don't use it.

Resolves: #1578

The Asterisk Development Team would like to announce
release candidate 2 of asterisk-21.12.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/21.12.0-rc2
and
https://downloads.asterisk.org/pub/telephony/asterisk

Repository: https://github.com/asterisk/asterisk
Tag: 21.12.0-rc2

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-rc2

Links:

• Full ChangeLog
• GitHub Diff
• Tarball
• Downloads

Summary:

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

User Notes:

Upgrade Notes:

Developer Notes:

Commit Authors:

• George Joseph: (1)

Issue and Commit Detail:

Closed Issues:

• 1578: [bug]: Deadlock with externalMedia custom channel id and cpp map channel backend

Commits By Author:

• George Joseph (1):

Commit List:

• channelstorage: Allow storage driver read locking to be skipped.

Commit Details:

channelstorage: Allow storage driver read locking to be skipped.

Author: George Joseph
Date: 2025-11-06

After PR #1498 added read locking to channelstorage_cpp_map_name_id, if ARI
channels/externalMedia was called with a custom channel id AND the
cpp_map_name_id channel storage backend is in use, a deadlock can occur when
hanging up the channel. It's actually triggered in
channel.c:__ast_channel_alloc_ap() when it gets a write lock on the
channelstorage driver then subsequently does a lookup for channel uniqueid
which now does a read lock. This is an invalid operation and causes the lock
state to get "bad". When the channels try to hang up, a write lock is
attempted again which hangs and causes the deadlock.

Now instead of the cpp_map_name_id channelstorage driver "get" APIs
automatically performing a read lock, they take a "lock" parameter which
allows a caller who already has a write lock to indicate that the "get" API
must not attempt its own lock. This prevents the state from getting mesed up.

The ao2_legacy driver uses the ao2 container's recursive mutex so doesn't
have this issue but since it also implements the common channelstorage API,
it needed its "get" implementations updated to take the lock parameter. They
just don't use it.

Resolves: #1578

The Asterisk Development Team would like to announce
release candidate 2 of asterisk-20.17.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/20.17.0-rc2
and
https://downloads.asterisk.org/pub/telephony/asterisk

Repository: https://github.com/asterisk/asterisk
Tag: 20.17.0-rc2

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-rc2

Links:

• Full ChangeLog
• GitHub Diff
• Tarball
• Downloads

Summary:

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

User Notes:

Upgrade Notes:

Developer Notes:

Commit Authors:

• George Joseph: (1)

Issue and Commit Detail:

Closed Issues:

• 1578: [bug]: Deadlock with externalMedia custom channel id and cpp map channel backend

Commits By Author:

• George Joseph (1):

Commit List:

• channelstorage: Allow storage driver read locking to be skipped.

Commit Details:

channelstorage: Allow storage driver read locking to be skipped.

Author: George Joseph
Date: 2025-11-06

After PR #1498 added read locking to channelstorage_cpp_map_name_id, if ARI
channels/externalMedia was called with a custom channel id AND the
cpp_map_name_id channel storage backend is in use, a deadlock can occur when
hanging up the channel. It's actually triggered in
channel.c:__ast_channel_alloc_ap() when it gets a write lock on the
channelstorage driver then subsequently does a lookup for channel uniqueid
which now does a read lock. This is an invalid operation and causes the lock
state to get "bad". When the channels try to hang up, a write lock is
attempted again which hangs and causes the deadlock.

Now instead of the cpp_map_name_id channelstorage driver "get" APIs
automatically performing a read lock, they take a "lock" parameter which
allows a caller who already has a write lock to indicate that the "get" API
must not attempt its own lock. This prevents the state from getting mesed up.

The ao2_legacy driver uses the ao2 container's recursive mutex so doesn't
have this issue but since it also implements the common channelstorage API,
it needed its "get" implementations updated to take the lock parameter. They
just don't use it.

Resolves: #1578

The Asterisk Development Team would like to announce
release candidate 1 of asterisk-22.7.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/22.7.0-rc1
and
https://downloads.asterisk.org/pub/telephony/asterisk

Repository: https://github.com/asterisk/asterisk
Tag: 22.7.0-rc1

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-rc1

Links:

• Full ChangeLog
• GitHub Diff
• Tarball
• Downloads

Summary:

• Commits: 53
• Commit Authors: 16
• Issues Resolved: 35
• 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:

• 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_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.

Commit Authors:

• Anthony Minessale: (1)
• Bastian Triller: (1)
• Ben Ford: (2)
• Christoph Moench-Tegeder: (1)
• Gauravs456: (1)
• George Joseph: (8)
• Igor Goncharovsky: (1)
• Joshua C. Colp: (8)
• Max Grobecker: (1)
• Nathan Monfils: (1)
• Naveen Albert: (18)
• Phoneben: (2)
• Roman Pertsev: (1)
• Sean Bright: (3)
• Sven Kube: (3)
• Tinet-Mucw: (1)

The Asterisk Development Team would like to announce
release candidate 1 of asterisk-23.1.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/23.1.0-rc1
and
https://downloads.asterisk.org/pub/telephony/asterisk

Repository: https://github.com/asterisk/asterisk
Tag: 23.1.0-rc1

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-rc1

Links:

• Full ChangeLog
• GitHub Diff
• Tarball
• Downloads

Summary:

• Commits: 54
• Commit Authors: 17
• 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:

• 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_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.

Commit Authors:

• Allan Nathanson: (1)
• Anthony Minessale: (1)
• Bastian Triller: (1)
• Ben Ford: (2)
• Christoph Moench-Tegeder: (1)
• Gauravs456: (1)
• George Joseph: (8)
• Igor Goncharovsky: (1)
• Joshua C. Colp: (8)
• Max Grobecker: (1)
• Nathan Monfils: (1)
• Naveen Albert: (18)
• Phoneben: (2)
• Roman Pertsev: (1)
• Sean Bright: (3)
• Sven Kube: (3)
• Tinet-Mucw: (1)