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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)

New uNmINeD development snapshot is available for download!

Changes:

• Fixed support for Bedrock worlds without a LevelDB log file
• Updated to .NET 10
• Updated to Avalonia 11.3.10

Download de Windows-versie voor 5 clients.

2025-12-23

nginx-1.28.1 stable version has been released.

There is a new PHP release in town!

💾

💾

💾

There is a new PHP release in town!

💾

💾

💾

There is a new PHP release in town!

💾

💾

💾

There is a new PHP release in town!

💾

💾

💾

There is a new PHP release in town!

💾

💾

💾

26.1 Snapshot 1 (known as 26.1-snapshot-1 in the launcher) is the first snapshot for Java Edition 26.1, released on December 16, 2025. It adds data pack-controlled trading, improves the game's lighting and the look of the Night Vision effect, and makes other technical changes (i.e. related to data packs). Full changelog: https://minecraft.wiki/Java_Edition_26.1-snapshot-1

Download de Windows-versie voor 5 clients.

2025-12-09

nginx-1.29.4 mainline version has been released, featuring HTTP/2 to backend and Encrypted ClientHello support.

2025-12-09

nginx-1.29.4 mainline version has been released, featuring HTTP/2 to backend and Encrypted ClientHello.

1.21.11, the release of Mounts of Mayhem, is a game drop for Java Edition released on December 9, 2025, which adds the nautilus, zombie nautilus, nautilus armor, spears, netherite horse armor, camel husks, parched, and zombie horsemen. Full changelog: https://minecraft.wiki/Java_Edition_1.21.11

Download de Windows-versie voor 5 clients.

1.21.11 Release Candidate 3 (known as 1.21.11-rc3 in the launcher) is the third and final release candidate for Java Edition 1.21.11, released on December 8, 2025. Full changelog: https://minecraft.wiki/Java_Edition_1.21.11-rc3

Download de Windows-versie voor 5 clients.

1.21.11 Release Candidate 2 (known as 1.21.11-rc2 in the launcher) is the second release candidate for Java Edition 1.21.11, released on December 5, 2025, which fixes a crash. Full changelog: https://minecraft.wiki/Java_Edition_1.21.11-rc2

1.21.11 Release Candidate 1 (known as 1.21.11-rc1 in the launcher) is the first release candidate for Java Edition 1.21.11, released on December 4, 2025, which tweaks spear mechanics. Full changelog: https://minecraft.wiki/Java_Edition_1.21.11-rc1

VIENNA, Austria – December 04, 2025 –Enterprise software developer Proxmox Server Solutions GmbH (henceforth “Proxmox”) today announced the immediate availability of the stable version 1.0 of Proxmox Datacenter Manager. This new product directly addresses the increasing complexity of operating distributed and large-scaled Proxmox-based environments. Proxmox Datacenter Manager offers a holistic single pane of glass view for the administration, monitoring, and scaling of Proxmox VE and Proxmox Backup Server, with the primary goal of providing administrators with comprehensive and seamless control.

Managing growing data centers, distributed across multiple locations or clusters, consistently presents major challenges for enterprises and teams. A lack of global oversight, fragmented metrics, and the need to perform complex operations manually across various environments can quickly lead to inefficiencies and increased error susceptibility.

Proxmox Datacenter Manager was developed as the strategic answer to this scaling challenge. It bridges the gap between individual Proxmox-based nodes and clusters, providing a unified view of the entire infrastructure. This not only simplifies routine tasks but also enables advanced functionalities that were previously difficult to achieve.

Highlights of Proxmox Datacenter Manager 1.0

Proxmox Datacenter Manager delivers a set of core functions specifically designed for managing complex, enterprise-grade environments:

• Centralized overview and metrics aggregation: Users can connect multiple Proxmox “remotes” (nodes and clusters) and gain a real-time, consolidated overview from a single dashboard. The consolidated dashboard displays the global health status of all Proxmox VE clusters and Proxmox Backup Server instances. It aggregates critical resource usage, including CPU, RAM, and storage I/O, and provides an immediate view of critical key performance indicators (KPIs) and performance metrics to identify bottlenecks and potential issues early on. Data is cached locally, maintaining offline visibility of the last known state.
• Dynamic, role-based custom views: With customizable dashboards, IT teams can create highly filtered, targeted overviews based on specific remotes, resource types, or operational tags. Crucially, the Proxmox Datacenter Manager leverages its native role-based access control (RBAC) to grant users access to these tailored views without providing direct access to the underlying virtual machines or hosts. This functionality ensures granular permission management and delivers need-to-know transparency across diverse teams and multi-tenant environments.
• Multi-cluster management: Seamlessly connect to and manage independent Proxmox-based clusters and standalone nodes.
• Cross-cluster live migration: One of the most prominent features is the capability for the live migration of VMs between different clusters. This empowers administrators to perform responsive load shifts and maintenance work without downtime.
• Basic VM & container life-cycle management for virtual infrastructure: Routine administrative tasks such as starting, stopping, or configuring VMs, containers, and storage resources can be executed directly from the central interface. Further, with the included native Role-Based Access Control (RBAC), Proxmox Datacenter Manager allows to precisely manage user permissions and centralize task histories and logs to simplify auditing and meeting compliance requirements.
• Powerful search functionality: Version 1.0 comes with a highly intuitive and powerful search functionality. Inspired by query languages like those used in Elasticsearch and GitHub, administrators can instantly filter and locate resources. Data can be filtered by resource type (remote, VM, container), status (stopped, running, etc.) or by custom tags, therefore ensuring that even in infrastructures managing thousands of virtual guests, critical resources and diagnostic data are found with unprecedented speed and precision.
• Centralized SDN capabilities (EVPN): The platform features support for Software-Defined Networking (SDN), enabling the configuration of EVPN zones and VNets across multiple remotes from a single interface, simplifying complex network overlays and network administration in highly scaled environments.
• Centralized update management: Proxmox Datacenter Manager introduces a central Update Management Panel that gives administrators an instant overview of all available updates across their entire Proxmox VE and Proxmox Backup Server infrastructure. Updates can be rolled out directly from the Datacenter Manager interface, simplifying patch management and strengthening the overall security posture. In addition, Datacenter Manager provides unified, secure shell access to all managed remotes from a single console.
• Open-source software stack: Proxmox Datacenter Manager is based on Debian 13.2 “Trixie”, uses a newer Linux kernel version 6.17 as stable default, and includes ZFS 2.3. Furthermore, its core software stack is written in the high-performance Rust programming language, with a responsive user interface built upon the new Rust/Yew Proxmox UI framework, delivering enhanced speed and an optimal user experience.

"The modern infrastructure landscape demands adaptability, from data centers to edge locations. Organizations need tools that evolve alongside their business. Proxmox Datacenter Manager is designed as a key building block within our expanding ecosystem, empowering customers with the right solution for every stage of their journey", says Tim Marx, COO at Proxmox. "By choosing the Proxmox ecosystem, organizations unlock a wide range of deployment options. From high-performance setups at hyperscalers to distributed branch offices that maintain data sovereignty. Our consistent commitment to openness ensures long-term interoperability and real freedom of choice for customers and partners."

Availability

Proxmox Datacenter Manager 1.0 is immediately available for download. Users can obtain a complete installation image via ISO download, which contains the full feature-set of the solution and can be installed quickly on bare-metal systems using an intuitive installation wizard.

Seamless distribution upgrades from older versions of Proxmox Datacenter Manager are possible using the standard APT package management system. Furthermore, it is also possible to install Proxmox Datacenter Manager on top of an existing Debian installation. As Free/Libre and Open Source Software (FLOSS), the entire solution is published under the GNU AGPLv3.

For enterprise users, Proxmox Server Solutions GmbH offers professional support through subscription plans. A subscription provides access to the stable Enterprise Repository with timely updates via the web interface, as well as to certified technical support and is recommended for production use. Customers with active Enterprise Support for their Proxmox remotes also gain access to Proxmox Datacenter Manager updates and support.

Resources:

• ISO Image Download: https://www.proxmox.com/en/downloads
• Forum Announcement: https://forum.proxmox.com/
• Roadmap: For published and upcoming features, see the Release Notes & Documentation

###

About Proxmox Server Solutions
Proxmox provides powerful and user-friendly open-source server software. Enterprises of all sizes and industries use the Proxmox solutions to deploy efficient and simplified IT infrastructures, minimize total cost of ownership, and avoid vendor lock-in. Proxmox also offers commercial support, training services, and an extensive partner ecosystem to ensure business continuity for its customers. Proxmox Server Solutions GmbH was established in 2005 and is headquartered in Vienna, Austria.

Contact: Daniela Häsler, Proxmox Server Solutions GmbH, marketing@proxmox.com