The Extended Stable channel has been updated to 142.0.7444.265 for Windows and Mac which will roll out over the coming days/weeks.
The Stable channel has been updated to 143.0.7499.192/.193 for Windows/Mac and 143.0.7499.192 for Linux, which will roll out over the coming days/weeks. A full list of changes in this build is available in the Log.
Security Fixes and Rewards
Note: Access to bug details and links may be kept restricted until a majority of users are updated with a fix. We will also retain restrictions if the bug exists in a third party library that other projects similarly depend on, but haven’t yet fixed.
Note: Access to bug details and links may be kept restricted until a majority of users are updated with a fix.
We will also retain restrictions if the bug exists in a third party library that other projects similarly depend on,
but haven’t yet fixed.
This update includes 1 security fix.
Below, we highlight fixes that were contributed by external researchers.
Please see the Chrome Security Page for more information.
[TBD][463155954] High CVE-2026-0628: Insufficient policy enforcement in WebView tag. Reported by Gal Weizman on 2025-11-23
We would also like to thank all security researchers that worked with us during the development cycle to prevent security bugs from ever reaching the stable channel.
Many of our security bugs are detected using AddressSanitizer, MemorySanitizer, UndefinedBehaviorSanitizer, Control Flow Integrity, libFuzzer, or AFL.
Interested in switching release channels? Find out how here. If you find a new issue, please let us know by filing a bug. The community help forum is also a great place to reach out for help or learn about common issues.
Fix issue #430: Autoformat on Save with empty XML-Elements.
Formatting Options has new setting to format Xml attributes each on a separate line.
Fix issue #425: "No byte order mark on save" option throws stream closed exception add unit test.
Apply dark mode to window titlebar.
Issue 409: Not able to validate XML against multiple not referenced Schemas
Issue 329: Incomplete schema validation of large XML files ( > ~20 MB).
Report invalid characters undex 0x20.
Fix unit tests so they run on Windows 11, including a work around for dotnet/winforms#10244.
Issue 324: Generate xml from schema, see ttps://youtu.be/5I_q1oXz02I.
Issue 385:: "XML Reload" removes loaded XSLT. You can now get auto-reload when xml file changes on disk
and you can turn off the prompt in options dialog and it will reapply the XSLT automatically.
update to WebView2 version 1.0.2739.15
Also preserve the selection in the XML tree view across file reloads.
Issue 382: Add progress bar on long XSLT transform operations.
Issue 379: Add Preserve Whitespace option so your XSLT transforms can output the whitespace correctly.
Bug 383: Reload of a "not loaded" file
Bug 381: Multiple instances resets the settings
Bug 380: Opening bad XML file keeps watching the file
Issue 382: Add progress bar on long XSLT transform operations.
Issue 379: Add Preserve Whitespace option so your XSLT transforms can output the whitespace correctly.
Bug 383: Reload of a "not loaded" file
Bug 381: Multiple instances resets the settings
Bug 380: Opening bad XML file keeps watching the file
Add ability to load .json files.
Add #367 preserve Options Dialog box relative location and size
Bug 370: Error Message after SEARCHING and drag & drop of a new file
Bug 371: Unhandled exception loading XML file with the http://www.crossref.org/schema/deposit/crossref4.4.2.xsd schemas.
Add ability to load .json files.
Add #367 preserve Options Dialog box relative location and size
Bug 370: Error Message after SEARCHING and drag & drop of a new file
Bug 371: Unhandled exception loading XML file with the http://www.crossref.org/schema/deposit/crossref4.4.2.xsd schemas.
No notable changes in this release. Bumping versions for a new TestFlight build.
Full Changelog: v1.7.0.8...v1.8.0.0
This release is only to provide an updated TestFlight build since the current build has expired.
There are no notable changes since the published 1.7.0.8 version.
Full Changelog: v1.7.0.8...v1.7.0.9
Tip
If you like Part-DB, consider donating to support the development. Press the sponsor button on the main github page, for more info.
Important
If you are using Part-DB it would be helpful if you fill out this short survey on your usage of Part-DB (Google Forms): https://forms.gle/Q15twx3YYq3qCNfe8
Full Changelog: v2.3.0...v2.4.0
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!
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.
CHANNEL(adsicpe) can now be read or written to change
the channels' ADSI CPE capability setting.
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.
The DIGIT_SUM function can be used to return the digit sum of
a number.
The 't' option for ReceiveSF now allows for a timer since
the last digit received, in addition to the number-wide timeout.
The answer and progress dial timeouts now have millisecond
precision, instead of having to be whole numbers.
The DAHDI_CHANNEL function allows for getting/setting
certain properties about DAHDI channels from the dialplan.
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.
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.
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.
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.
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.
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.
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!
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.
The DAHDI_CHANNEL function allows for getting/setting
certain properties about DAHDI channels from the dialplan.
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!
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.
CHANNEL(adsicpe) can now be read or written to change
the channels' ADSI CPE capability setting.
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.
The DIGIT_SUM function can be used to return the digit sum of
a number.
The 't' option for ReceiveSF now allows for a timer since
the last digit received, in addition to the number-wide timeout.
The answer and progress dial timeouts now have millisecond
precision, instead of having to be whole numbers.
The DAHDI_CHANNEL function allows for getting/setting
certain properties about DAHDI channels from the dialplan.
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.
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.
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.
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.
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.
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.
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!
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.
CHANNEL(adsicpe) can now be read or written to change
the channels' ADSI CPE capability setting.
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.
The DIGIT_SUM function can be used to return the digit sum of
a number.
The 't' option for ReceiveSF now allows for a timer since
the last digit received, in addition to the number-wide timeout.
The answer and progress dial timeouts now have millisecond
precision, instead of having to be whole numbers.
The DAHDI_CHANNEL function allows for getting/setting
certain properties about DAHDI channels from the dialplan.
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.
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.
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.
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.
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.
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!
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!
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!
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!
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!
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.
CHANNEL(adsicpe) can now be read or written to change
the channels' ADSI CPE capability setting.
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.
The DIGIT_SUM function can be used to return the digit sum of
a number.
The 't' option for ReceiveSF now allows for a timer since
the last digit received, in addition to the number-wide timeout.
The answer and progress dial timeouts now have millisecond
precision, instead of having to be whole numbers.
The DAHDI_CHANNEL function allows for getting/setting
certain properties about DAHDI channels from the dialplan.
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.
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.
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.
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.
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.
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.
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.
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!
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.
CHANNEL(adsicpe) can now be read or written to change
the channels' ADSI CPE capability setting.
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.
The DIGIT_SUM function can be used to return the digit sum of
a number.
The 't' option for ReceiveSF now allows for a timer since
the last digit received, in addition to the number-wide timeout.
The answer and progress dial timeouts now have millisecond
precision, instead of having to be whole numbers.
The DAHDI_CHANNEL function allows for getting/setting
certain properties about DAHDI channels from the dialplan.
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.
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.
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.
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.
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.
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.
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.
Log in.deb - (e95aca8)Package variable - (13c55ce)v4 of git-cliff-action - (81d613b)build-* - (16a54a5)tx-push recipe - (d293d64).ini files on run-* - (c513e53)run-gtk2, run-qt5 - (04650d9)tx-push to Makefile - (3ceed39)secrets.mk - (123a2e7)Note
For a list of all the changes up to date, please read CHANGELOG.md.
Note: You can already get a preview of v13 for Windows, built on the same codebase as the Linux version. See the
download page for a zip package.
3rd party updates:
Bugfixes and enhancements:
build-* - (16a54a5)tx-push recipe - (d293d64).ini files on run-* - (c513e53)run-gtk2, run-qt5 - (04650d9)tx-push to Makefile - (3ceed39)secrets.mk - (123a2e7)Note
For a list of all the changes up to date, please read CHANGELOG.md.
Tip
This version has problems loading libmariadb library. The issue will be fixed in the next release. To work around in this release you may switch to libmysqlclient instead. If the library drop-down does not show libmysqlclient, install it via sudo apt install libmysqlclient-dev.
Get it from the download page
3rd party updates:
