The official Asterisk Project repository.
The Asterisk Development Team would like to announce
the release of asterisk-21.2.0.
The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/21.2.0
and
https://downloads.asterisk.org/pub/telephony/asterisk
Repository: https://github.com/asterisk/asterisk Tag: 21.2.0
This release resolves issues reported by the community
and would have not been possible without your participation.
Thank You!
The timeout argument to Dial now allows specifying the maximum amount of time to dial if early media is not received.
The leaveurgent mailbox option can now be used to control whether callers may leave messages marked as 'Urgent'.
Asterisk's stir-shaken feature has been refactored to correct interoperability, RFC compliance, and performance issues. See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
Bundled pjproject has been upgraded to 2.14. For more information on what all is included in this change, check out the pjproject Github page: https://github.com/pjsip/pjproject/releases
PJSIP outbound registrations now support a per-registration User-Agent header
The SpeechBackground dialplan application now supports a 'p' option that will return partial results from speech engines that provide them when a timeout occurs.
The ChanSpy application now accepts the 'D' option which will interleave the spied audio within the outgoing frames. The purpose of this is to allow the audio to be read as a Dual channel stream with separate incoming and outgoing audio. Setting both the 'o' option and the 'D' option and results in the 'D' option being ignored.
The fix requires removing the macrocontext column from the voicemail_messages table in the voicemail database via alembic upgrade.
The 'dahdi set mwi' now allows MWI on channels to be manually toggled if needed for troubleshooting. Resolves: #440
The stir-shaken refactor is a breaking change but since it's not working now we don't think it matters. The stir_shaken.conf file has changed significantly which means that existing ones WILL need to be changed. The stir_shaken.conf.sample file in configs/samples/ has quite a bit more information. This is also an ABI breaking change since some of the existing objects needed to be changed or removed, and new ones added. Additionally, if res_stir_shaken is enabled in menuselect, you'll need to either have the development package for libjwt v1.15.3 installed or use the --with-libjwt-bundled option with ./configure.
The fix requires that the voicemail database be upgraded via alembic. Upgrading to the latest voicemail database via alembic will remove the macrocontext column from the voicemail_messages table.
The Asterisk Development Team would like to announce
the release of asterisk-20.7.0.
The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/20.7.0
and
https://downloads.asterisk.org/pub/telephony/asterisk
Repository: https://github.com/asterisk/asterisk Tag: 20.7.0
This release resolves issues reported by the community
and would have not been possible without your participation.
Thank You!
The timeout argument to Dial now allows specifying the maximum amount of time to dial if early media is not received.
The leaveurgent mailbox option can now be used to control whether callers may leave messages marked as 'Urgent'.
Asterisk's stir-shaken feature has been refactored to correct interoperability, RFC compliance, and performance issues. See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
Bundled pjproject has been upgraded to 2.14. For more information on what all is included in this change, check out the pjproject Github page: https://github.com/pjsip/pjproject/releases
The SpeechBackground dialplan application now supports a 'p' option that will return partial results from speech engines that provide them when a timeout occurs.
The ChanSpy application now accepts the 'D' option which will interleave the spied audio within the outgoing frames. The purpose of this is to allow the audio to be read as a Dual channel stream with separate incoming and outgoing audio. Setting both the 'o' option and the 'D' option and results in the 'D' option being ignored.
The 'dahdi set mwi' now allows MWI on channels to be manually toggled if needed for troubleshooting. Resolves: #440
The Asterisk Development Team would like to announce
the release of asterisk-18.22.0.
The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/18.22.0
and
https://downloads.asterisk.org/pub/telephony/asterisk
Repository: https://github.com/asterisk/asterisk Tag: 18.22.0
This release resolves issues reported by the community
and would have not been possible without your participation.
Thank You!
The timeout argument to Dial now allows specifying the maximum amount of time to dial if early media is not received.
The leaveurgent mailbox option can now be used to control whether callers may leave messages marked as 'Urgent'.
Asterisk's stir-shaken feature has been refactored to correct interoperability, RFC compliance, and performance issues. See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
Bundled pjproject has been upgraded to 2.14. For more information on what all is included in this change, check out the pjproject Github page: https://github.com/pjsip/pjproject/releases
The SpeechBackground dialplan application now supports a 'p' option that will return partial results from speech engines that provide them when a timeout occurs.
The ChanSpy application now accepts the 'D' option which will interleave the spied audio within the outgoing frames. The purpose of this is to allow the audio to be read as a Dual channel stream with separate incoming and outgoing audio. Setting both the 'o' option and the 'D' option and results in the 'D' option being ignored.
The 'dahdi set mwi' now allows MWI on channels to be manually toggled if needed for troubleshooting. Resolves: #440
The Asterisk Development Team would like to announce
the release of Certified asterisk-18.9-cert8.
The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/certified-18.9-cert8
and
https://downloads.asterisk.org/pub/telephony/certified-asterisk
Repository: https://github.com/asterisk/asterisk Tag: certified-18.9-cert8
This release resolves issues reported by the community
and would have not been possible without your participation.
Thank You!
The ChanSpy application now accepts the 'D' option which will interleave the spied audio within the outgoing frames. The purpose of this is to allow the audio to be read as a Dual channel stream with separate incoming and outgoing audio. Setting both the 'o' option and the 'D' option and results in the 'D' option being ignored.
A new dialplan app PJSIPHangup and AMI action allows you to hang up an unanswered incoming PJSIP call with a specific SIP response code in the 400 -> 699 range.
res_speech now supports translation of an input channel to a format supported by the speech provider, provided a translation path is available between the source format and provider capabilites.
Call setup times should be significantly improved when using ARI.
You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS. This removes a significant amount of overhead if you just want to detect possible deadlocks vs needing full lock tracing.
A new option "sounds_search_custom_dir" has been added to
asterisk.conf that allows asterisk to search
AST_DATA_DIR/sounds/custom for sounds files before searching the
standard AST_DATA_DIR/sounds/
The "Build Options" entry in the "core show settings" CLI command has been renamed to "ABI related Build Options" and a new entry named "All Build Options" has been added that shows both breaking and non-breaking options.
Four new dialplan functions have been added. GLOBAL_DELETE and DELETE have been added which allows the deletion of global and channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been added which checks whether a global or channel variable has been set.
The 'queue priority caller' CLI command and 'QueueChangePriorityCaller' AMI action now have an 'immediate' argument which allows the caller priority change to be reflected immediately, causing the position of a caller to move within the queue depending on the priorities of the other callers.
The following manager actions have been added VoicemailBoxSummary - Generate message list for a given mailbox VoicemailRemove - Remove a message from a mailbox folder VoicemailMove - Move a message from one folder to another within a mailbox VoicemailForward - Copy a message from one folder in one mailbox to another folder in another or the same mailbox.
The following CLI commands have been added to app_voicemail
voicemail show mailbox
The Asterisk Development Team would like to announce
release candidate 2 of asterisk-21.2.0.
The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/21.2.0-rc2
and
https://downloads.asterisk.org/pub/telephony/asterisk
Repository: https://github.com/asterisk/asterisk Tag: 21.2.0-rc2
This release resolves issues reported by the community
and would have not been possible without your participation.
Thank You!
The Asterisk Development Team would like to announce
release candidate 2 of asterisk-20.7.0.
The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/20.7.0-rc2
and
https://downloads.asterisk.org/pub/telephony/asterisk
Repository: https://github.com/asterisk/asterisk Tag: 20.7.0-rc2
This release resolves issues reported by the community
and would have not been possible without your participation.
Thank You!
The Asterisk Development Team would like to announce
release candidate 2 of asterisk-18.22.0.
The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/18.22.0-rc2
and
https://downloads.asterisk.org/pub/telephony/asterisk
Repository: https://github.com/asterisk/asterisk Tag: 18.22.0-rc2
This release resolves issues reported by the community
and would have not been possible without your participation.
Thank You!
The Asterisk Development Team would like to announce
release candidate 1 of asterisk-21.2.0.
The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/21.2.0-rc1
and
https://downloads.asterisk.org/pub/telephony/asterisk
Repository: https://github.com/asterisk/asterisk Tag: 21.2.0-rc1
This release resolves issues reported by the community
and would have not been possible without your participation.
Thank You!
The timeout argument to Dial now allows specifying the maximum amount of time to dial if early media is not received.
The leaveurgent mailbox option can now be used to control whether callers may leave messages marked as 'Urgent'.
Asterisk's stir-shaken feature has been refactored to correct interoperability, RFC compliance, and performance issues. See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
Bundled pjproject has been upgraded to 2.14. For more information on what all is included in this change, check out the pjproject Github page: https://github.com/pjsip/pjproject/releases
PJSIP outbound registrations now support a per-registration User-Agent header
The SpeechBackground dialplan application now supports a 'p' option that will return partial results from speech engines that provide them when a timeout occurs.
The ChanSpy application now accepts the 'D' option which will interleave the spied audio within the outgoing frames. The purpose of this is to allow the audio to be read as a Dual channel stream with separate incoming and outgoing audio. Setting both the 'o' option and the 'D' option and results in the 'D' option being ignored.
The fix requires removing the macrocontext column from the voicemail_messages table in the voicemail database via alembic upgrade.
The 'dahdi set mwi' now allows MWI on channels to be manually toggled if needed for troubleshooting. Resolves: #440
The stir-shaken refactor is a breaking change but since it's not working now we don't think it matters. The stir_shaken.conf file has changed significantly which means that existing ones WILL need to be changed. The stir_shaken.conf.sample file in configs/samples/ has quite a bit more information. This is also an ABI breaking change since some of the existing objects needed to be changed or removed, and new ones added. Additionally, if res_stir_shaken is enabled in menuselect, you'll need to either have the development package for libjwt v1.15.3 installed or use the --with-libjwt-bundled option with ./configure.
The fix requires that the voicemail database be upgraded via alembic. Upgrading to the latest voicemail database via alembic will remove the macrocontext column from the voicemail_messages table.
The Asterisk Development Team would like to announce
release candidate 1 of asterisk-20.7.0.
The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/20.7.0-rc1
and
https://downloads.asterisk.org/pub/telephony/asterisk
Repository: https://github.com/asterisk/asterisk Tag: 20.7.0-rc1
This release resolves issues reported by the community
and would have not been possible without your participation.
Thank You!
The timeout argument to Dial now allows specifying the maximum amount of time to dial if early media is not received.
The leaveurgent mailbox option can now be used to control whether callers may leave messages marked as 'Urgent'.
Asterisk's stir-shaken feature has been refactored to correct interoperability, RFC compliance, and performance issues. See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
Bundled pjproject has been upgraded to 2.14. For more information on what all is included in this change, check out the pjproject Github page: https://github.com/pjsip/pjproject/releases
The SpeechBackground dialplan application now supports a 'p' option that will return partial results from speech engines that provide them when a timeout occurs.
The ChanSpy application now accepts the 'D' option which will interleave the spied audio within the outgoing frames. The purpose of this is to allow the audio to be read as a Dual channel stream with separate incoming and outgoing audio. Setting both the 'o' option and the 'D' option and results in the 'D' option being ignored.
The 'dahdi set mwi' now allows MWI on channels to be manually toggled if needed for troubleshooting. Resolves: #440