❌

Lees weergave

Asterisk Release 22.7.0

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:

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)

  •  

Asterisk Release 21.12.0

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:

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)

  •  

Asterisk Release 23.1.0

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:

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)

  •  

Asterisk Release 20.17.0

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:

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)

  •  

Asterisk Release 23.1.0-rc2

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:

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

  •  

Asterisk Release 22.7.0-rc2

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:

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

  •  

Asterisk Release 21.12.0-rc2

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:

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

  •  

Asterisk Release 20.17.0-rc2

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:

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

  •  

Asterisk Release 22.7.0-rc1

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:

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)

  •  

Asterisk Release 23.1.0-rc1

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:

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)

  •  
❌