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Published by asterisk-org-access-app[bot] 3 months ago
The Asterisk Development Team would like to announce
release candidate 1 of asterisk-18.24.0.
The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/18.24.0-rc1
and
https://downloads.asterisk.org/pub/telephony/asterisk
Repository: https://github.com/asterisk/asterisk
Tag: 18.24.0-rc1
This release resolves issues reported by the community
and would have not been possible without your participation.
Thank You!
This commit adds a new voicemail.conf option
'odbc_audio_on_disk' which when set causes the ODBC variant of
app_voicemail_odbc to leave the message and greeting audio files
on disk and only store the message metadata in the database.
Much more information can be found in the voicemail.conf.sample
file.
Add a Queue option log-restricted-caller-id to control whether the Restricted Caller ID
will be stored in the queue log.
If log-restricted-caller-id=no then the Caller ID will be stripped if the Caller ID is restricted.
The fields width of "core show hints" were increased.
The width of "extension" field to 30 characters and
the width of the "device state id" field to 60 characters.
No change in configuration is required in order to enable this
feature. Endpoints configured to use RFC2833 will automatically have this
enabled. If the endpoint does not support this, it should not include it in
the SDP offer/response.
Resolves: #699
Published by asterisk-org-access-app[bot] 5 months ago
The Asterisk Development Team would like to announce security release
Asterisk 21.3.1.
The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/21.3.1
and
https://downloads.asterisk.org/pub/telephony/asterisk
Repository: https://github.com/asterisk/asterisk
Tag: 21.3.1
Published by asterisk-org-access-app[bot] 5 months ago
The Asterisk Development Team would like to announce security release
Asterisk 20.8.1.
The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/20.8.1
and
https://downloads.asterisk.org/pub/telephony/asterisk
Repository: https://github.com/asterisk/asterisk
Tag: 20.8.1
Published by asterisk-org-access-app[bot] 5 months ago
The Asterisk Development Team would like to announce security release
Asterisk 18.23.1.
The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/18.23.1
and
https://downloads.asterisk.org/pub/telephony/asterisk
Repository: https://github.com/asterisk/asterisk
Tag: 18.23.1
Published by asterisk-org-access-app[bot] 5 months ago
The Asterisk Development Team would like to announce
the release of asterisk-21.3.0.
The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/21.3.0
and
https://downloads.asterisk.org/pub/telephony/asterisk
Repository: https://github.com/asterisk/asterisk
Tag: 21.3.0
This release resolves issues reported by the community
and would have not been possible without your participation.
Thank You!
Issuing "pjsip reload" will no longer disable
logging if it was previously enabled from the CLI.
In certain circumstances, modules with dependency relations
can have their dependents automatically recursively unloaded and loaded
again using the "module refresh" CLI command or the ModuleLoad AMI command.
Secure websocket client connections now send SNI in
the TLS client hello.
set identify_by=transport for the pjsip endpoint. Then
use the existing 'match' option and the new 'transport' option of
the identify.
Fixes: #672
this new feature let users match endpoints based on the
indound SIP requests' URI. To do so, add 'request_uri' to the
endpoint's 'identify_by' option. The 'match_request_uri' option of
the identify can be an exact match for the entire request uri, or a
regular expression (between slashes). It's quite similar to the
header identifer.
Fixes: #599
the GET_TRANSFERRER_DATA dialplan variable can now be used also in pjsip.
When using the Originate AMI Action, we now can pass the PreDialGoSub parameter, instructing the asterisk to perform an subrouting at channel before call start. With this parameter an call initiated by AMI can request the channel to start the call automaticaly, adding a SIP header to using GoSUB, instructing to autoanswer the channel, and proceeding the outbuound extension executing. Exemple of an context to perform the previus indication:
[addautoanswer]
exten => _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
exten => _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
exten => _s,n,Return()
The "manager kick session" CLI command now
allows kicking a specified AMI session.
"waitfordialtone" may now be specified for DAHDI
trunk channels on a per-call basis using the CHANNEL function.
Bundled pjproject has been upgraded to 2.14.1. For more
information visit pjproject Github page: https://github.com/pjsip/pjproject/releases/tag/2.14.1
Published by asterisk-org-access-app[bot] 5 months ago
The Asterisk Development Team would like to announce
the release of asterisk-20.8.0.
The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/20.8.0
and
https://downloads.asterisk.org/pub/telephony/asterisk
Repository: https://github.com/asterisk/asterisk
Tag: 20.8.0
This release resolves issues reported by the community
and would have not been possible without your participation.
Thank You!
Issuing "pjsip reload" will no longer disable
logging if it was previously enabled from the CLI.
In certain circumstances, modules with dependency relations
can have their dependents automatically recursively unloaded and loaded
again using the "module refresh" CLI command or the ModuleLoad AMI command.
Secure websocket client connections now send SNI in
the TLS client hello.
set identify_by=transport for the pjsip endpoint. Then
use the existing 'match' option and the new 'transport' option of
the identify.
Fixes: #672
this new feature let users match endpoints based on the
indound SIP requests' URI. To do so, add 'request_uri' to the
endpoint's 'identify_by' option. The 'match_request_uri' option of
the identify can be an exact match for the entire request uri, or a
regular expression (between slashes). It's quite similar to the
header identifer.
Fixes: #599
the GET_TRANSFERRER_DATA dialplan variable can now be used also in pjsip.
When using the Originate AMI Action, we now can pass the PreDialGoSub parameter, instructing the asterisk to perform an subrouting at channel before call start. With this parameter an call initiated by AMI can request the channel to start the call automaticaly, adding a SIP header to using GoSUB, instructing to autoanswer the channel, and proceeding the outbuound extension executing. Exemple of an context to perform the previus indication:
[addautoanswer]
exten => _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
exten => _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
exten => _s,n,Return()
The "manager kick session" CLI command now
allows kicking a specified AMI session.
"waitfordialtone" may now be specified for DAHDI
trunk channels on a per-call basis using the CHANNEL function.
Bundled pjproject has been upgraded to 2.14.1. For more
information visit pjproject Github page: https://github.com/pjsip/pjproject/releases/tag/2.14.1
Published by asterisk-org-access-app[bot] 5 months ago
The Asterisk Development Team would like to announce
the release of asterisk-18.23.0.
The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/18.23.0
and
https://downloads.asterisk.org/pub/telephony/asterisk
Repository: https://github.com/asterisk/asterisk
Tag: 18.23.0
This release resolves issues reported by the community
and would have not been possible without your participation.
Thank You!
Issuing "pjsip reload" will no longer disable
logging if it was previously enabled from the CLI.
In certain circumstances, modules with dependency relations
can have their dependents automatically recursively unloaded and loaded
again using the "module refresh" CLI command or the ModuleLoad AMI command.
Secure websocket client connections now send SNI in
the TLS client hello.
set identify_by=transport for the pjsip endpoint. Then
use the existing 'match' option and the new 'transport' option of
the identify.
Fixes: #672
this new feature let users match endpoints based on the
indound SIP requests' URI. To do so, add 'request_uri' to the
endpoint's 'identify_by' option. The 'match_request_uri' option of
the identify can be an exact match for the entire request uri, or a
regular expression (between slashes). It's quite similar to the
header identifer.
Fixes: #599
the GET_TRANSFERRER_DATA dialplan variable can now be used also in pjsip.
When using the Originate AMI Action, we now can pass the PreDialGoSub parameter, instructing the asterisk to perform an subrouting at channel before call start. With this parameter an call initiated by AMI can request the channel to start the call automaticaly, adding a SIP header to using GoSUB, instructing to autoanswer the channel, and proceeding the outbuound extension executing. Exemple of an context to perform the previus indication:
[addautoanswer]
exten => _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
exten => _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
exten => _s,n,Return()
The "manager kick session" CLI command now
allows kicking a specified AMI session.
"waitfordialtone" may now be specified for DAHDI
trunk channels on a per-call basis using the CHANNEL function.
Bundled pjproject has been upgraded to 2.14.1. For more
information visit pjproject Github page: https://github.com/pjsip/pjproject/releases/tag/2.14.1
Published by asterisk-org-access-app[bot] 5 months ago
The Asterisk Development Team would like to announce
release candidate 2 of Certified asterisk-20.7-cert1.
The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert1-rc2
and
https://downloads.asterisk.org/pub/telephony/certified-asterisk
Repository: https://github.com/asterisk/asterisk
Tag: certified-20.7-cert1-rc2
This release resolves issues reported by the community
and would have not been possible without your participation.
Thank You!
Published by asterisk-org-access-app[bot] 5 months ago
The Asterisk Development Team would like to announce
the release of Certified asterisk-18.9-cert9.
The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/certified-18.9-cert9
and
https://downloads.asterisk.org/pub/telephony/certified-asterisk
Repository: https://github.com/asterisk/asterisk
Tag: certified-18.9-cert9
This release resolves issues reported by the community
and would have not been possible without your participation.
Thank You!
Published by asterisk-org-access-app[bot] 5 months ago
The Asterisk Development Team would like to announce
release candidate 1 of asterisk-20.8.0.
The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/20.8.0-rc1
and
https://downloads.asterisk.org/pub/telephony/asterisk
Repository: https://github.com/asterisk/asterisk
Tag: 20.8.0-rc1
This release resolves issues reported by the community
and would have not been possible without your participation.
Thank You!
Issuing "pjsip reload" will no longer disable
logging if it was previously enabled from the CLI.
In certain circumstances, modules with dependency relations
can have their dependents automatically recursively unloaded and loaded
again using the "module refresh" CLI command or the ModuleLoad AMI command.
Secure websocket client connections now send SNI in
the TLS client hello.
set identify_by=transport for the pjsip endpoint. Then
use the existing 'match' option and the new 'transport' option of
the identify.
Fixes: #672
this new feature let users match endpoints based on the
indound SIP requests' URI. To do so, add 'request_uri' to the
endpoint's 'identify_by' option. The 'match_request_uri' option of
the identify can be an exact match for the entire request uri, or a
regular expression (between slashes). It's quite similar to the
header identifer.
Fixes: #599
the GET_TRANSFERRER_DATA dialplan variable can now be used also in pjsip.
When using the Originate AMI Action, we now can pass the PreDialGoSub parameter, instructing the asterisk to perform an subrouting at channel before call start. With this parameter an call initiated by AMI can request the channel to start the call automaticaly, adding a SIP header to using GoSUB, instructing to autoanswer the channel, and proceeding the outbuound extension executing. Exemple of an context to perform the previus indication:
[addautoanswer]
exten => _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
exten => _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
exten => _s,n,Return()
The "manager kick session" CLI command now
allows kicking a specified AMI session.
"waitfordialtone" may now be specified for DAHDI
trunk channels on a per-call basis using the CHANNEL function.
Bundled pjproject has been upgraded to 2.14.1. For more
information visit pjproject Github page: https://github.com/pjsip/pjproject/releases/tag/2.14.1
Published by asterisk-org-access-app[bot] 5 months ago
The Asterisk Development Team would like to announce
release candidate 1 of asterisk-21.3.0.
The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/21.3.0-rc1
and
https://downloads.asterisk.org/pub/telephony/asterisk
Repository: https://github.com/asterisk/asterisk
Tag: 21.3.0-rc1
This release resolves issues reported by the community
and would have not been possible without your participation.
Thank You!
Issuing "pjsip reload" will no longer disable
logging if it was previously enabled from the CLI.
In certain circumstances, modules with dependency relations
can have their dependents automatically recursively unloaded and loaded
again using the "module refresh" CLI command or the ModuleLoad AMI command.
Secure websocket client connections now send SNI in
the TLS client hello.
set identify_by=transport for the pjsip endpoint. Then
use the existing 'match' option and the new 'transport' option of
the identify.
Fixes: #672
this new feature let users match endpoints based on the
indound SIP requests' URI. To do so, add 'request_uri' to the
endpoint's 'identify_by' option. The 'match_request_uri' option of
the identify can be an exact match for the entire request uri, or a
regular expression (between slashes). It's quite similar to the
header identifer.
Fixes: #599
the GET_TRANSFERRER_DATA dialplan variable can now be used also in pjsip.
When using the Originate AMI Action, we now can pass the PreDialGoSub parameter, instructing the asterisk to perform an subrouting at channel before call start. With this parameter an call initiated by AMI can request the channel to start the call automaticaly, adding a SIP header to using GoSUB, instructing to autoanswer the channel, and proceeding the outbuound extension executing. Exemple of an context to perform the previus indication:
[addautoanswer]
exten => _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
exten => _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
exten => _s,n,Return()
The "manager kick session" CLI command now
allows kicking a specified AMI session.
"waitfordialtone" may now be specified for DAHDI
trunk channels on a per-call basis using the CHANNEL function.
Bundled pjproject has been upgraded to 2.14.1. For more
information visit pjproject Github page: https://github.com/pjsip/pjproject/releases/tag/2.14.1
Published by asterisk-org-access-app[bot] 5 months ago
The Asterisk Development Team would like to announce
release candidate 1 of asterisk-18.23.0.
The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/18.23.0-rc1
and
https://downloads.asterisk.org/pub/telephony/asterisk
Repository: https://github.com/asterisk/asterisk
Tag: 18.23.0-rc1
This release resolves issues reported by the community
and would have not been possible without your participation.
Thank You!
Issuing "pjsip reload" will no longer disable
logging if it was previously enabled from the CLI.
In certain circumstances, modules with dependency relations
can have their dependents automatically recursively unloaded and loaded
again using the "module refresh" CLI command or the ModuleLoad AMI command.
Secure websocket client connections now send SNI in
the TLS client hello.
set identify_by=transport for the pjsip endpoint. Then
use the existing 'match' option and the new 'transport' option of
the identify.
Fixes: #672
this new feature let users match endpoints based on the
indound SIP requests' URI. To do so, add 'request_uri' to the
endpoint's 'identify_by' option. The 'match_request_uri' option of
the identify can be an exact match for the entire request uri, or a
regular expression (between slashes). It's quite similar to the
header identifer.
Fixes: #599
the GET_TRANSFERRER_DATA dialplan variable can now be used also in pjsip.
When using the Originate AMI Action, we now can pass the PreDialGoSub parameter, instructing the asterisk to perform an subrouting at channel before call start. With this parameter an call initiated by AMI can request the channel to start the call automaticaly, adding a SIP header to using GoSUB, instructing to autoanswer the channel, and proceeding the outbuound extension executing. Exemple of an context to perform the previus indication:
[addautoanswer]
exten => _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
exten => _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
exten => _s,n,Return()
The "manager kick session" CLI command now
allows kicking a specified AMI session.
"waitfordialtone" may now be specified for DAHDI
trunk channels on a per-call basis using the CHANNEL function.
Bundled pjproject has been upgraded to 2.14.1. For more
information visit pjproject Github page: https://github.com/pjsip/pjproject/releases/tag/2.14.1
Published by asterisk-org-access-app[bot] 7 months ago
The Asterisk Development Team would like to announce
release candidate 1 of Certified asterisk-20.7-cert1.
The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert1-rc1
and
https://downloads.asterisk.org/pub/telephony/certified-asterisk
Repository: https://github.com/asterisk/asterisk
Tag: certified-20.7-cert1-rc1
This release resolves issues reported by the community
and would have not been possible without your participation.
Thank You!
rtp->themssrc_valid
i..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 option "j" is now available for the Dial application which
uses the initial stream topology of the caller to create the outgoing
channels.
The console log can now be filtered by
channels or groups of channels, using the
logger filter CLI commands.
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.
The VoicemailPasswordChange event is
now emitted whenever a mailbox password is updated,
containing the mailbox information and the new
password.
Resolves: #398
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.
With this update, the PJSIP realm lengths have been extended
to support up to 255 characters.
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/ directory.
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.
The dial string option 'g' was added to the UnicastRTP channel
which enables RTP glue and therefore native RTP bridges with those
channels.
Introduce a new queue configuration option called
'periodic-announce-startdelay' which will vary the normal (historic)
behavior of starting the periodic announcement cycle at
periodic-announce-frequency seconds after entering the queue to start
the periodic announcement cycle at period-announce-startdelay seconds
after joining the queue. The default behavior if this config option is
not set remains unchanged.
Signed-off-by: Jaco Kroon [email protected]
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.
Called Subscriber Held is now supported for analog
FXS channels, using the calledsubscriberheld option. This allows
a station user to go on hook when receiving an incoming call
and resume from another phone on the same line by going on hook,
without disconnecting the call.
The prefix argument to PJSIP_HEADERS is now
optional. If not specified, all header names will be
returned.
There is a new ARI endpoint /endpoints/refer
for referring
an endpoint to some URI or endpoint.
The autoreoriginate setting now allows for kewlstart FXS
channels to automatically reoriginate and provide dial tone to the
user again after all calls on the line have cleared. This saves users
from having to manually hang up and pick up the receiver again before
making another call.
The threewaysilenthold option now allows the three-way
dial tone to time out to silence, rather than continuing forever.
res_pjsip now allows TLS v1.3 to be enabled if supported by
the underlying PJSIP library. The bundled version of PJSIP supports
TLS v1.3.
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
Show contents of mailbox @
voicemail remove <from_folder>
Remove message from <from_folder> in mailbox @
voicemail move <from_folder> <to_folder>
Move message in mailbox & from <from_folder> to <to_folder>
voicemail forward <from_mailbox> <from_context> <from_folder> <to_mailbox> <to_context> <to_folder>
Forward message in mailbox @ <from_folder> to
mailbox @ <to_folder>
The immediatering option can now be set to no to suppress
the fake audible ringback provided when immediate=yes on FXS channels.
New ParkingSpace parameter has been added to AMI action Park.
The loop_last option in musiconhold.conf now
allows the last file in the directory to be looped once reached.
New AMI action CoreShowChannelMap has been added.
Additional Caller ID properties are now supported on
incoming calls to FXS stations, namely the
redirecting reason and call qualifier.
When creating a bridge using the ARI the 'type' argument now
accepts a new value 'sdp_label' which will configure the bridge to add
labels for each stream in the SDP with the corresponding channel id.
Make paused reason in realtime queues persist an
Asterisk restart. This was fixed for non-realtime
queues in ASTERISK_25732.
The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used
by itself or in conert with the existing
AST_CEL_LOCAL_OPTIMIZE to book-end local channel optimizaion.
A "dialmode" option has been added which allows
specifying, on a per-channel basis, what methods of
subscriber dialing (pulse and/or tone) are permitted.
Additionally, this can be changed on a channel
at any point during a call using the CHANNEL
function.
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.
Ampersands in URLs passed to the Playback()
,
Background()
, SpeechBackground()
, Read()
, Authenticate()
, or
Queue()
applications as filename arguments can now be escaped by
single quoting the filename. Additionally, this is also possible when
using the CONFBRIDGE
dialplan function, or configuring various
features in confbridge.conf
and queues.conf
.
The dtls_rekey will be disabled if webrtc support is
requested on an endpoint. A warning will also be emitted.
As part of this update, the maximum allowable length
for PJSIP endpoints and relevant resources has been increased from
40 to 255 characters. To take advantage of this enhancement, it is
recommended to run the necessary procedures (e.g., Alembic) to
update your schemas.
Add a new column to the queue_member table:
reason_paused VARCHAR(80) so the reason can be preserved.
The existing AST_CEL_LOCAL_OPTIMIZE can continue
to be used as-is and the AST_CEL_LOCAL_OPTIMIZE_BEGIN event
can be ignored if desired.
An additional 751 ASTERISK-* issues were closed.
Published by asterisk-org-access-app[bot] 7 months ago
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.
Published by asterisk-org-access-app[bot] 7 months ago
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
Published by asterisk-org-access-app[bot] 7 months ago
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
Published by asterisk-org-access-app[bot] 7 months ago
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/ directory.
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
Show contents of mailbox @
voicemail remove <from_folder>
Remove message from <from_folder> in mailbox @
voicemail move <from_folder> <to_folder>
Move message in mailbox & from <from_folder> to <to_folder>
voicemail forward <from_mailbox> <from_context> <from_folder> <to_mailbox> <to_context> <to_folder>
Forward message in mailbox @ <from_folder> to
mailbox @ <to_folder>
Published by asterisk-org-access-app[bot] 7 months ago
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!
Published by asterisk-org-access-app[bot] 7 months ago
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!
Published by asterisk-org-access-app[bot] 7 months ago
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!