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asterisk - Asterisk Release 20.5.1

Published by asterisk-org-access-app[bot] 10 months ago

The Asterisk Development Team would like to announce security release
Asterisk 20.5.1.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/20.5.1
and
https://downloads.asterisk.org/pub/telephony/asterisk

The following security advisories were resolved in this release:

Change Log for Release asterisk-20.5.1

Summary:

res_pjsip_header_funcs: Duplicate new header value, don't copy.
res_pjsip: disable raw bad packet logging
res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
manager.c: Prevent path traversal with GetConfig.

User Notes:

Upgrade Notes:

Closed Issues:

None

asterisk - Asterisk Release 18.20.1

Published by asterisk-org-access-app[bot] 10 months ago

The Asterisk Development Team would like to announce security release
Asterisk 18.20.1.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/18.20.1
and
https://downloads.asterisk.org/pub/telephony/asterisk

The following security advisories were resolved in this release:

Change Log for Release asterisk-18.20.1

Summary:

res_pjsip_header_funcs: Duplicate new header value, don't copy.
res_pjsip: disable raw bad packet logging
res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
manager.c: Prevent path traversal with GetConfig.

User Notes:

Upgrade Notes:

Closed Issues:

None

asterisk - Asterisk Release 21.0.0

Published by asterisk-org-access-app[bot] about 1 year ago

The Asterisk Development Team would like to announce
the release of asterisk-21.0.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/21.0.0
and
https://downloads.asterisk.org/pub/telephony/asterisk

This release resolves issues reported by the community
and would have not been possible without your participation.

Thank You!

Change Log for Release asterisk-21.0.0

Summary:

Update master branch for Asterisk 21
translate.c: Prefer better codecs upon translate ties.
chan_skinny: Remove deprecated module.
app_osplookup: Remove deprecated module.
chan_mgcp: Remove deprecated module.
chan_alsa: Remove deprecated module.
pbx_builtins: Remove deprecated and defunct functionality.
chan_sip: Remove deprecated module.
app_cdr: Remove deprecated application and option.
app_macro: Remove deprecated module.
res_monitor: Remove deprecated module.
http.c: Minor simplification to HTTP status output.
app_osplookup: Remove obsolete sample config.
say.c: Fix French time playback. (#42)
core: Cleanup gerrit and JIRA references. (#58)
utils.h: Deprecate ast_gethostbyname(). (#79)
res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
app_sla: Migrate SLA applications out of app_meetme.
rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
.github: Update AsteriskReleaser for security releases
users.conf: Deprecate users.conf configuration.
Update version for Asterisk 21
Remove unneeded CHANGES and UPGRADE files
res_pjsip_pubsub: Add body_type to test_handler for unit tests
ari-stubs: Fix more local anchor references
ari-stubs: Fix more local anchor references
ari-stubs: Fix broken documentation anchors
res_pjsip_session: Send Session Interval too small response
.github: Update workflow-application-token-action to v2
app_dial: Fix infinite loop when sending digits.
app_voicemail: Fix for loop declarations
alembic: Fix quoting of the 100rel column
pbx.c: Fix gcc 12 compiler warning.
app_audiosocket: Fixed timeout with -1 to avoid busy loop.
download_externals: Fix a few version related issues
main/refer.c: Fix double free in refer_data_destructor + potential leak
sig_analog: Add Called Subscriber Held capability.
Revert "app_stack: Print proper exit location for PBXless channels."
install_prereq: Fix dependency install on aarch64.
res_pjsip.c: Set contact_user on incoming call local Contact header
extconfig: Allow explicit DB result set ordering to be disabled.
rest-api: Run make ari-stubs
res_pjsip_header_funcs: Make prefix argument optional.
pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
manager: Tolerate stasis messages with no channel snapshot.
Remove unneeded CHANGES and UPGRADE files

User Notes:

sig_analog: Add Called Subscriber Held capability.

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.
res_pjsip_header_funcs: Make prefix argument optional.

The prefix argument to PJSIP_HEADERS is now
optional. If not specified, all header names will be
returned.
http.c: Minor simplification to HTTP status output.

For bound addresses, the HTTP status page now combines the bound
address and bound port in a single line. Additionally, the SSL bind
address has been renamed to TLS.

Upgrade Notes:

utils.h: Deprecate ast_gethostbyname(). (#79)

ast_gethostbyname() has been deprecated and will be removed
in Asterisk 23. New code should use ast_sockaddr_resolve() and
ast_sockaddr_resolve_first_af().
app_sla: Migrate SLA applications out of app_meetme.

The SLAStation and SLATrunk applications have been moved
from app_meetme to app_sla. If you are using these applications and have
autoload=no, you will need to explicitly load this module in modules.conf.
users.conf: Deprecate users.conf configuration.

The users.conf config is now deprecated
and will be removed in a future version of Asterisk.
res_monitor: Remove deprecated module.

This module was deprecated in Asterisk 16
and is now being removed in accordance with
the Asterisk Module Deprecation policy.
This also removes the 'w' and 'W' options
for app_queue.
MixMonitor should be default and only option
for all settings that previously used either
Monitor or MixMonitor.
app_osplookup: Remove deprecated module.

This module was deprecated in Asterisk 19
and is now being removed in accordance with
the Asterisk Module Deprecation policy.
app_cdr: Remove deprecated application and option.

The previously deprecated NoCDR application has been removed.
Additionally, the previously deprecated 'e' option to the ResetCDR
application has been removed.
app_macro: Remove deprecated module.

This module was deprecated in Asterisk 16
and is now being removed in accordance with
the Asterisk Module Deprecation policy.
For most modules that interacted with app_macro,
this change is limited to no longer looking for
the current context from the macrocontext when set.
The following modules have additional impacts:
app_dial - no longer supports M^ connected/redirecting macro
app_minivm - samples written using macro will no longer work.
The sample needs to be re-written
app_queue - can no longer call a macro on the called party's
channel. Use gosub which is currently supported
ccss - no callback macro, gosub only
app_voicemail - no macro support
channel - remove macrocontext and priority, no connected
line or redirection macro options
options - stdexten is deprecated to gosub as the default
and only options
pbx - removed macrolock
pbx_dundi - no longer look for macro
snmp - removed macro context, exten, and priority
translate.c: Prefer better codecs upon translate ties.

When setting up translation between two codecs the quality was not taken into account,
resulting in suboptimal translation. The quality is now taken into account,
which can reduce the number of translation steps required, and improve the resulting quality.
chan_sip: Remove deprecated module.

This module was deprecated in Asterisk 17
and is now being removed in accordance with
the Asterisk Module Deprecation policy.
chan_alsa: Remove deprecated module.

This module was deprecated in Asterisk 19
and is now being removed in accordance with
the Asterisk Module Deprecation policy.
pbx_builtins: Remove deprecated and defunct functionality.

The previously deprecated ImportVar and SetAMAFlags
applications have now been removed.
chan_mgcp: Remove deprecated module.

This module was deprecated in Asterisk 19
and is now being removed in accordance with
the Asterisk Module Deprecation policy.
chan_skinny: Remove deprecated module.

This module was deprecated in Asterisk 19
and is now being removed in accordance with
the Asterisk Module Deprecation policy.

Closed Issues:

#37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on aarch64 Debian platforms
#39: [Bug]: Remove .gitreview from repository.
#41: [Bug]: say.c Time announcement does not say o'clock for the French language
#50: [improvement]: app_sla: Migrate SLA applications from app_meetme
#78: [improvement]: Deprecate ast_gethostbyname()
#81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
#179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
#183: [deprecation]: Deprecate users.conf
#226: [improvement]: Apply contact_user to incoming calls
#230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
#240: [new-feature]: sig_analog: Add Called Subscriber Held capability
#253: app_gosub patch appear to have broken predial handlers that utilize macros that call gosubs
#255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules registered"
#263: [bug]: download_externals doesn't always handle versions correctly
#267: [bug]: ari: refer with display_name key in request body leads to crash
#274: [bug]: Syntax Error in SQL Code
#275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99 -Wdeclaration-after-statement'
#277: [bug]: pbx.c: Compiler error with gcc 12.2
#281: [bug]: app_dial: Infinite loop if called channel hangs up while receiving digits
#335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in build_resource_tree

asterisk - Asterisk Release 20.5.0

Published by asterisk-org-access-app[bot] about 1 year ago

The Asterisk Development Team would like to announce
the release of asterisk-20.5.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/20.5.0
and
https://downloads.asterisk.org/pub/telephony/asterisk

This release resolves issues reported by the community
and would have not been possible without your participation.

Thank You!

Change Log for Release asterisk-20.5.0

Summary:

ari-stubs: Fix more local anchor references
ari-stubs: Fix more local anchor references
ari-stubs: Fix broken documentation anchors
res_pjsip_session: Send Session Interval too small response
.github: Update workflow-application-token-action to v2
app_dial: Fix infinite loop when sending digits.
app_voicemail: Fix for loop declarations
alembic: Fix quoting of the 100rel column
pbx.c: Fix gcc 12 compiler warning.
app_audiosocket: Fixed timeout with -1 to avoid busy loop.
download_externals: Fix a few version related issues
main/refer.c: Fix double free in refer_data_destructor + potential leak
sig_analog: Add Called Subscriber Held capability.
app_macro: Fix locking around datastore access
Revert "app_stack: Print proper exit location for PBXless channels."
.github: Use generic releaser
install_prereq: Fix dependency install on aarch64.
res_pjsip.c: Set contact_user on incoming call local Contact header
extconfig: Allow explicit DB result set ordering to be disabled.
rest-api: Run make ari-stubs
res_pjsip_header_funcs: Make prefix argument optional.
pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
manager: Tolerate stasis messages with no channel snapshot.
core/ari/pjsip: Add refer mechanism
chan_dahdi: Allow autoreoriginating after hangup.
audiohook: Unlock channel in mute if no audiohooks present.
sig_analog: Allow three-way flash to time out to silence.
res_prometheus: Do not generate broken metrics
res_pjsip: Enable TLS v1.3 if present.
func_cut: Add example to documentation.
extensions.conf.sample: Remove reference to missing context.
func_export: Use correct function argument as variable name.
app_queue: Add support for applying caller priority change immediately.
.github: Fix cherry-pick reminder issues
chan_iax2.c: Avoid crash with IAX2 switch support.
res_geolocation: Ensure required 'location_info' is present.
Adds manager actions to allow move/remove/forward individual messages in a particular mailbox folder. The forward command can be used to copy a message within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry, required to retrieve message ID's.
app_voicemail: add CLI commands for message manipulation
res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using rtp->themssrc_valid into the scope of the rtp_instance lock.
.github: Minor tweak to Asterisk Releaser
.github: Suppress cherry-pick reminder for some situations
sig_analog: Allow immediate fake ring to be suppressed.

User Notes:

sig_analog: Add Called Subscriber Held capability.

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.
res_pjsip_header_funcs: Make prefix argument optional.

The prefix argument to PJSIP_HEADERS is now
optional. If not specified, all header names will be
returned.
core/ari/pjsip: Add refer mechanism

There is a new ARI endpoint /endpoints/refer for referring
an endpoint to some URI or endpoint.
chan_dahdi: Allow autoreoriginating after hangup.

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.
sig_analog: Allow three-way flash to time out to silence.

The threewaysilenthold option now allows the three-way
dial tone to time out to silence, rather than continuing forever.
res_pjsip: Enable TLS v1.3 if present.

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.
app_queue: Add support for applying caller priority change immediately.

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.
Adds manager actions to allow move/remove/forward individual messages in a particular mailbox folder. The forward command can be used to copy a message within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry, required to retrieve message ID's.

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.
app_voicemail: add CLI commands for message manipulation

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>
sig_analog: Allow immediate fake ring to be suppressed.

The immediatering option can now be set to no to suppress
the fake audible ringback provided when immediate=yes on FXS channels.

Upgrade Notes:

Closed Issues:

#37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on aarch64 Debian platforms
#71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to some resource
#118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when immediate=yes
#170: [improvement]: app_voicemail - add CLI commands to manipulate messages
#179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
#181: [improvement]: app_voicemail - add manager actions to display and manipulate messages
#202: [improvement]: app_queue: Add support for immediately applying queue caller priority change
#205: [new-feature]: sig_analog: Allow flash to time out to silent hold
#224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
#226: [improvement]: Apply contact_user to incoming calls
#230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
#233: [bug]: Deadlock with MixMonitorMute AMI action
#240: [new-feature]: sig_analog: Add Called Subscriber Held capability
#253: app_gosub patch appear to have broken predial handlers that utilize macros that call gosubs
#255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules registered"
#263: [bug]: download_externals doesn't always handle versions correctly
#265: [bug]: app_macro isn't locking around channel datastore access
#267: [bug]: ari: refer with display_name key in request body leads to crash
#274: [bug]: Syntax Error in SQL Code
#275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99 -Wdeclaration-after-statement'
#277: [bug]: pbx.c: Compiler error with gcc 12.2
#281: [bug]: app_dial: Infinite loop if called channel hangs up while receiving digits

asterisk - Asterisk Release 18.20.0

Published by asterisk-org-access-app[bot] about 1 year ago

The Asterisk Development Team would like to announce
the release of asterisk-18.20.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/18.20.0
and
https://downloads.asterisk.org/pub/telephony/asterisk

This release resolves issues reported by the community
and would have not been possible without your participation.

Thank You!

Change Log for Release asterisk-18.20.0

Summary:

ari-stubs: Fix more local anchor references
ari-stubs: Fix more local anchor references
ari-stubs: Fix broken documentation anchors
res_pjsip_session: Send Session Interval too small response
.github: Update workflow-application-token-action to v2
app_dial: Fix infinite loop when sending digits.
app_voicemail: Fix for loop declarations
alembic: Fix quoting of the 100rel column
pbx.c: Fix gcc 12 compiler warning.
app_audiosocket: Fixed timeout with -1 to avoid busy loop.
download_externals: Fix a few version related issues
main/refer.c: Fix double free in refer_data_destructor + potential leak
sig_analog: Add Called Subscriber Held capability.
app_macro: Fix locking around datastore access
Revert "app_stack: Print proper exit location for PBXless channels."
.github: Use generic releaser
install_prereq: Fix dependency install on aarch64.
res_pjsip.c: Set contact_user on incoming call local Contact header
extconfig: Allow explicit DB result set ordering to be disabled.
rest-api: Run make ari-stubs
res_pjsip_header_funcs: Make prefix argument optional.
pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
manager: Tolerate stasis messages with no channel snapshot.
core/ari/pjsip: Add refer mechanism
chan_dahdi: Allow autoreoriginating after hangup.
audiohook: Unlock channel in mute if no audiohooks present.
sig_analog: Allow three-way flash to time out to silence.
res_prometheus: Do not generate broken metrics
res_pjsip: Enable TLS v1.3 if present.
func_cut: Add example to documentation.
extensions.conf.sample: Remove reference to missing context.
func_export: Use correct function argument as variable name.
app_queue: Add support for applying caller priority change immediately.
.github: Fix cherry-pick reminder issues
chan_iax2.c: Avoid crash with IAX2 switch support.
res_geolocation: Ensure required 'location_info' is present.
Adds manager actions to allow move/remove/forward individual messages in a particular mailbox folder. The forward command can be used to copy a message within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry, required to retrieve message ID's.
app_voicemail: add CLI commands for message manipulation
res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using rtp->themssrc_valid into the scope of the rtp_instance lock.
.github: Minor tweak to Asterisk Releaser
.github: Suppress cherry-pick reminder for some situations
sig_analog: Allow immediate fake ring to be suppressed.

User Notes:

sig_analog: Add Called Subscriber Held capability.

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.
res_pjsip_header_funcs: Make prefix argument optional.

The prefix argument to PJSIP_HEADERS is now
optional. If not specified, all header names will be
returned.
core/ari/pjsip: Add refer mechanism

There is a new ARI endpoint /endpoints/refer for referring
an endpoint to some URI or endpoint.
chan_dahdi: Allow autoreoriginating after hangup.

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.
sig_analog: Allow three-way flash to time out to silence.

The threewaysilenthold option now allows the three-way
dial tone to time out to silence, rather than continuing forever.
res_pjsip: Enable TLS v1.3 if present.

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.
app_queue: Add support for applying caller priority change immediately.

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.
Adds manager actions to allow move/remove/forward individual messages in a particular mailbox folder. The forward command can be used to copy a message within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry, required to retrieve message ID's.

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.
app_voicemail: add CLI commands for message manipulation

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>
sig_analog: Allow immediate fake ring to be suppressed.

The immediatering option can now be set to no to suppress
the fake audible ringback provided when immediate=yes on FXS channels.

Upgrade Notes:

Closed Issues:

#37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on aarch64 Debian platforms
#71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to some resource
#118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when immediate=yes
#170: [improvement]: app_voicemail - add CLI commands to manipulate messages
#179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
#181: [improvement]: app_voicemail - add manager actions to display and manipulate messages
#202: [improvement]: app_queue: Add support for immediately applying queue caller priority change
#205: [new-feature]: sig_analog: Allow flash to time out to silent hold
#224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
#226: [improvement]: Apply contact_user to incoming calls
#230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
#233: [bug]: Deadlock with MixMonitorMute AMI action
#240: [new-feature]: sig_analog: Add Called Subscriber Held capability
#253: app_gosub patch appear to have broken predial handlers that utilize macros that call gosubs
#255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules registered"
#263: [bug]: download_externals doesn't always handle versions correctly
#265: [bug]: app_macro isn't locking around channel datastore access
#267: [bug]: ari: refer with display_name key in request body leads to crash
#274: [bug]: Syntax Error in SQL Code
#275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99 -Wdeclaration-after-statement'
#277: [bug]: pbx.c: Compiler error with gcc 12.2
#281: [bug]: app_dial: Infinite loop if called channel hangs up while receiving digits

asterisk - Asterisk Release 21.0.0-rc1

Published by asterisk-org-access-app[bot] about 1 year ago

The Asterisk Development Team would like to announce
release candidate 1 of asterisk-21.0.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/21.0.0-rc1
and
https://downloads.asterisk.org/pub/telephony/asterisk

This release resolves issues reported by the community
and would have not been possible without your participation.

Thank You!

Change Log for Release asterisk-21.0.0-rc1

Summary:

Update master branch for Asterisk 21
translate.c: Prefer better codecs upon translate ties.
chan_skinny: Remove deprecated module.
app_osplookup: Remove deprecated module.
chan_mgcp: Remove deprecated module.
chan_alsa: Remove deprecated module.
pbx_builtins: Remove deprecated and defunct functionality.
chan_sip: Remove deprecated module.
app_cdr: Remove deprecated application and option.
app_macro: Remove deprecated module.
res_monitor: Remove deprecated module.
http.c: Minor simplification to HTTP status output.
app_osplookup: Remove obsolete sample config.
say.c: Fix French time playback. (#42)
core: Cleanup gerrit and JIRA references. (#58)
utils.h: Deprecate ast_gethostbyname(). (#79)
res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
app_sla: Migrate SLA applications out of app_meetme.
Update config.yml
rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
.github: Update AsteriskReleaser for security releases
users.conf: Deprecate users.conf configuration.
Update version for Asterisk 21
Remove unneeded CHANGES and UPGRADE files
ari-stubs: Fix more local anchor references
ari-stubs: Fix more local anchor references
ari-stubs: Fix broken documentation anchors
res_pjsip_session: Send Session Interval too small response
.github: Update workflow-application-token-action to v2
app_dial: Fix infinite loop when sending digits.
app_voicemail: Fix for loop declarations
alembic: Fix quoting of the 100rel column
pbx.c: Fix gcc 12 compiler warning.
app_audiosocket: Fixed timeout with -1 to avoid busy loop.
download_externals: Fix a few version related issues
main/refer.c: Fix double free in refer_data_destructor + potential leak
sig_analog: Add Called Subscriber Held capability.
Revert "app_stack: Print proper exit location for PBXless channels."
install_prereq: Fix dependency install on aarch64.
res_pjsip.c: Set contact_user on incoming call local Contact header
extconfig: Allow explicit DB result set ordering to be disabled.
rest-api: Run make ari-stubs
res_pjsip_header_funcs: Make prefix argument optional.
pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
manager: Tolerate stasis messages with no channel snapshot.
Remove unneeded CHANGES and UPGRADE files

User Notes:

sig_analog: Add Called Subscriber Held capability.

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.
res_pjsip_header_funcs: Make prefix argument optional.

The prefix argument to PJSIP_HEADERS is now
optional. If not specified, all header names will be
returned.
http.c: Minor simplification to HTTP status output.

For bound addresses, the HTTP status page now combines the bound
address and bound port in a single line. Additionally, the SSL bind
address has been renamed to TLS.

Upgrade Notes:

utils.h: Deprecate ast_gethostbyname(). (#79)

ast_gethostbyname() has been deprecated and will be removed
in Asterisk 23. New code should use ast_sockaddr_resolve() and
ast_sockaddr_resolve_first_af().
app_sla: Migrate SLA applications out of app_meetme.

The SLAStation and SLATrunk applications have been moved
from app_meetme to app_sla. If you are using these applications and have
autoload=no, you will need to explicitly load this module in modules.conf.
users.conf: Deprecate users.conf configuration.

The users.conf config is now deprecated
and will be removed in a future version of Asterisk.
app_osplookup: Remove deprecated module.

This module was deprecated in Asterisk 19
and is now being removed in accordance with
the Asterisk Module Deprecation policy.
res_monitor: Remove deprecated module.

This module was deprecated in Asterisk 16
and is now being removed in accordance with
the Asterisk Module Deprecation policy.
This also removes the 'w' and 'W' options
for app_queue.
MixMonitor should be default and only option
for all settings that previously used either
Monitor or MixMonitor.
chan_sip: Remove deprecated module.

This module was deprecated in Asterisk 17
and is now being removed in accordance with
the Asterisk Module Deprecation policy.
chan_alsa: Remove deprecated module.

This module was deprecated in Asterisk 19
and is now being removed in accordance with
the Asterisk Module Deprecation policy.
chan_mgcp: Remove deprecated module.

This module was deprecated in Asterisk 19
and is now being removed in accordance with
the Asterisk Module Deprecation policy.
chan_skinny: Remove deprecated module.

This module was deprecated in Asterisk 19
and is now being removed in accordance with
the Asterisk Module Deprecation policy.
app_macro: Remove deprecated module.

This module was deprecated in Asterisk 16
and is now being removed in accordance with
the Asterisk Module Deprecation policy.
For most modules that interacted with app_macro,
this change is limited to no longer looking for
the current context from the macrocontext when set.
The following modules have additional impacts:
app_dial - no longer supports M^ connected/redirecting macro
app_minivm - samples written using macro will no longer work.
The sample needs to be re-written
app_queue - can no longer call a macro on the called party's
channel. Use gosub which is currently supported
ccss - no callback macro, gosub only
app_voicemail - no macro support
channel - remove macrocontext and priority, no connected
line or redirection macro options
options - stdexten is deprecated to gosub as the default
and only options
pbx - removed macrolock
pbx_dundi - no longer look for macro
snmp - removed macro context, exten, and priority
pbx_builtins: Remove deprecated and defunct functionality.

The previously deprecated ImportVar and SetAMAFlags
applications have now been removed.
translate.c: Prefer better codecs upon translate ties.

When setting up translation between two codecs the quality was not taken into account,
resulting in suboptimal translation. The quality is now taken into account,
which can reduce the number of translation steps required, and improve the resulting quality.
app_cdr: Remove deprecated application and option.

The previously deprecated NoCDR application has been removed.
Additionally, the previously deprecated 'e' option to the ResetCDR
application has been removed.

Closed Issues:

#37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on aarch64 Debian platforms
#39: [Bug]: Remove .gitreview from repository.
#41: [Bug]: say.c Time announcement does not say o'clock for the French language
#50: [improvement]: app_sla: Migrate SLA applications from app_meetme
#78: [improvement]: Deprecate ast_gethostbyname()
#81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
#179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
#183: [deprecation]: Deprecate users.conf
#226: [improvement]: Apply contact_user to incoming calls
#230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
#240: [new-feature]: sig_analog: Add Called Subscriber Held capability
#253: app_gosub patch appear to have broken predial handlers that utilize macros that call gosubs
#255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules registered"
#263: [bug]: download_externals doesn't always handle versions correctly
#267: [bug]: ari: refer with display_name key in request body leads to crash
#274: [bug]: Syntax Error in SQL Code
#275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99 -Wdeclaration-after-statement'
#277: [bug]: pbx.c: Compiler error with gcc 12.2
#281: [bug]: app_dial: Infinite loop if called channel hangs up while receiving digits

asterisk - Asterisk Release 21.0.0-rc1

Published by asterisk-org-access-app[bot] about 1 year ago

The Asterisk Development Team would like to announce
release candidate 1 of asterisk-21.0.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/21.0.0-rc1
and
https://downloads.asterisk.org/pub/telephony/asterisk

This release resolves issues reported by the community
and would have not been possible without your participation.

Thank You!

Change Log for Release asterisk-21.0.0-rc1

Summary:

Update master branch for Asterisk 21
translate.c: Prefer better codecs upon translate ties.
chan_skinny: Remove deprecated module.
app_osplookup: Remove deprecated module.
chan_mgcp: Remove deprecated module.
chan_alsa: Remove deprecated module.
pbx_builtins: Remove deprecated and defunct functionality.
chan_sip: Remove deprecated module.
app_cdr: Remove deprecated application and option.
app_macro: Remove deprecated module.
res_monitor: Remove deprecated module.
http.c: Minor simplification to HTTP status output.
app_osplookup: Remove obsolete sample config.
say.c: Fix French time playback. (#42)
core: Cleanup gerrit and JIRA references. (#58)
utils.h: Deprecate ast_gethostbyname(). (#79)
res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
app_sla: Migrate SLA applications out of app_meetme.
Update config.yml
rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
.github: Update AsteriskReleaser for security releases
users.conf: Deprecate users.conf configuration.
Update version for Asterisk 21
Remove unneeded CHANGES and UPGRADE files
ari-stubs: Fix more local anchor references
ari-stubs: Fix more local anchor references
ari-stubs: Fix broken documentation anchors
res_pjsip_session: Send Session Interval too small response
.github: Update workflow-application-token-action to v2
app_dial: Fix infinite loop when sending digits.
app_voicemail: Fix for loop declarations
alembic: Fix quoting of the 100rel column
pbx.c: Fix gcc 12 compiler warning.
app_audiosocket: Fixed timeout with -1 to avoid busy loop.
download_externals: Fix a few version related issues
main/refer.c: Fix double free in refer_data_destructor + potential leak
sig_analog: Add Called Subscriber Held capability.
Revert "app_stack: Print proper exit location for PBXless channels."
install_prereq: Fix dependency install on aarch64.
res_pjsip.c: Set contact_user on incoming call local Contact header
extconfig: Allow explicit DB result set ordering to be disabled.
rest-api: Run make ari-stubs
res_pjsip_header_funcs: Make prefix argument optional.
pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
manager: Tolerate stasis messages with no channel snapshot.
Remove unneeded CHANGES and UPGRADE files

User Notes:

sig_analog: Add Called Subscriber Held capability.

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.
res_pjsip_header_funcs: Make prefix argument optional.

The prefix argument to PJSIP_HEADERS is now
optional. If not specified, all header names will be
returned.
http.c: Minor simplification to HTTP status output.

For bound addresses, the HTTP status page now combines the bound
address and bound port in a single line. Additionally, the SSL bind
address has been renamed to TLS.

Upgrade Notes:

utils.h: Deprecate ast_gethostbyname(). (#79)

ast_gethostbyname() has been deprecated and will be removed
in Asterisk 23. New code should use ast_sockaddr_resolve() and
ast_sockaddr_resolve_first_af().
app_sla: Migrate SLA applications out of app_meetme.

The SLAStation and SLATrunk applications have been moved
from app_meetme to app_sla. If you are using these applications and have
autoload=no, you will need to explicitly load this module in modules.conf.
users.conf: Deprecate users.conf configuration.

The users.conf config is now deprecated
and will be removed in a future version of Asterisk.
app_osplookup: Remove deprecated module.

This module was deprecated in Asterisk 19
and is now being removed in accordance with
the Asterisk Module Deprecation policy.
res_monitor: Remove deprecated module.

This module was deprecated in Asterisk 16
and is now being removed in accordance with
the Asterisk Module Deprecation policy.
This also removes the 'w' and 'W' options
for app_queue.
MixMonitor should be default and only option
for all settings that previously used either
Monitor or MixMonitor.
chan_sip: Remove deprecated module.

This module was deprecated in Asterisk 17
and is now being removed in accordance with
the Asterisk Module Deprecation policy.
chan_alsa: Remove deprecated module.

This module was deprecated in Asterisk 19
and is now being removed in accordance with
the Asterisk Module Deprecation policy.
chan_mgcp: Remove deprecated module.

This module was deprecated in Asterisk 19
and is now being removed in accordance with
the Asterisk Module Deprecation policy.
chan_skinny: Remove deprecated module.

This module was deprecated in Asterisk 19
and is now being removed in accordance with
the Asterisk Module Deprecation policy.
app_macro: Remove deprecated module.

This module was deprecated in Asterisk 16
and is now being removed in accordance with
the Asterisk Module Deprecation policy.
For most modules that interacted with app_macro,
this change is limited to no longer looking for
the current context from the macrocontext when set.
The following modules have additional impacts:
app_dial - no longer supports M^ connected/redirecting macro
app_minivm - samples written using macro will no longer work.
The sample needs to be re-written
app_queue - can no longer call a macro on the called party's
channel. Use gosub which is currently supported
ccss - no callback macro, gosub only
app_voicemail - no macro support
channel - remove macrocontext and priority, no connected
line or redirection macro options
options - stdexten is deprecated to gosub as the default
and only options
pbx - removed macrolock
pbx_dundi - no longer look for macro
snmp - removed macro context, exten, and priority
pbx_builtins: Remove deprecated and defunct functionality.

The previously deprecated ImportVar and SetAMAFlags
applications have now been removed.
translate.c: Prefer better codecs upon translate ties.

When setting up translation between two codecs the quality was not taken into account,
resulting in suboptimal translation. The quality is now taken into account,
which can reduce the number of translation steps required, and improve the resulting quality.
app_cdr: Remove deprecated application and option.

The previously deprecated NoCDR application has been removed.
Additionally, the previously deprecated 'e' option to the ResetCDR
application has been removed.

Closed Issues:

#37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on aarch64 Debian platforms
#39: [Bug]: Remove .gitreview from repository.
#41: [Bug]: say.c Time announcement does not say o'clock for the French language
#50: [improvement]: app_sla: Migrate SLA applications from app_meetme
#78: [improvement]: Deprecate ast_gethostbyname()
#81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
#179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
#183: [deprecation]: Deprecate users.conf
#226: [improvement]: Apply contact_user to incoming calls
#230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
#240: [new-feature]: sig_analog: Add Called Subscriber Held capability
#253: app_gosub patch appear to have broken predial handlers that utilize macros that call gosubs
#255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules registered"
#263: [bug]: download_externals doesn't always handle versions correctly
#267: [bug]: ari: refer with display_name key in request body leads to crash
#274: [bug]: Syntax Error in SQL Code
#275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99 -Wdeclaration-after-statement'
#277: [bug]: pbx.c: Compiler error with gcc 12.2
#281: [bug]: app_dial: Infinite loop if called channel hangs up while receiving digits

asterisk - Asterisk Release 20.5.0-rc1

Published by asterisk-org-access-app[bot] about 1 year ago

The Asterisk Development Team would like to announce
release candidate 1 of asterisk-20.5.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/20.5.0-rc1
and
https://downloads.asterisk.org/pub/telephony/asterisk

This release resolves issues reported by the community
and would have not been possible without your participation.

Thank You!

Change Log for Release asterisk-20.5.0-rc1

Summary:

ari-stubs: Fix more local anchor references
ari-stubs: Fix more local anchor references
ari-stubs: Fix broken documentation anchors
res_pjsip_session: Send Session Interval too small response
.github: Update workflow-application-token-action to v2
app_dial: Fix infinite loop when sending digits.
app_voicemail: Fix for loop declarations
alembic: Fix quoting of the 100rel column
pbx.c: Fix gcc 12 compiler warning.
app_audiosocket: Fixed timeout with -1 to avoid busy loop.
download_externals: Fix a few version related issues
main/refer.c: Fix double free in refer_data_destructor + potential leak
sig_analog: Add Called Subscriber Held capability.
app_macro: Fix locking around datastore access
Revert "app_stack: Print proper exit location for PBXless channels."
.github: Use generic releaser
install_prereq: Fix dependency install on aarch64.
res_pjsip.c: Set contact_user on incoming call local Contact header
extconfig: Allow explicit DB result set ordering to be disabled.
rest-api: Run make ari-stubs
res_pjsip_header_funcs: Make prefix argument optional.
pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
manager: Tolerate stasis messages with no channel snapshot.
core/ari/pjsip: Add refer mechanism
chan_dahdi: Allow autoreoriginating after hangup.
audiohook: Unlock channel in mute if no audiohooks present.
sig_analog: Allow three-way flash to time out to silence.
res_prometheus: Do not generate broken metrics
res_pjsip: Enable TLS v1.3 if present.
func_cut: Add example to documentation.
extensions.conf.sample: Remove reference to missing context.
func_export: Use correct function argument as variable name.
app_queue: Add support for applying caller priority change immediately.
.github: Fix cherry-pick reminder issues
chan_iax2.c: Avoid crash with IAX2 switch support.
res_geolocation: Ensure required 'location_info' is present.
Adds manager actions to allow move/remove/forward individual messages in a particular mailbox folder. The forward command can be used to copy a message within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry, required to retrieve message ID's.
app_voicemail: add CLI commands for message manipulation
res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using rtp->themssrc_valid into the scope of the rtp_instance lock.
.github: Minor tweak to Asterisk Releaser
.github: Suppress cherry-pick reminder for some situations
sig_analog: Allow immediate fake ring to be suppressed.

User Notes:

sig_analog: Add Called Subscriber Held capability.

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.
res_pjsip_header_funcs: Make prefix argument optional.

The prefix argument to PJSIP_HEADERS is now
optional. If not specified, all header names will be
returned.
core/ari/pjsip: Add refer mechanism

There is a new ARI endpoint /endpoints/refer for referring
an endpoint to some URI or endpoint.
chan_dahdi: Allow autoreoriginating after hangup.

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.
sig_analog: Allow three-way flash to time out to silence.

The threewaysilenthold option now allows the three-way
dial tone to time out to silence, rather than continuing forever.
res_pjsip: Enable TLS v1.3 if present.

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.
app_queue: Add support for applying caller priority change immediately.

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.
Adds manager actions to allow move/remove/forward individual messages in a particular mailbox folder. The forward command can be used to copy a message within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry, required to retrieve message ID's.

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.
app_voicemail: add CLI commands for message manipulation

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>
sig_analog: Allow immediate fake ring to be suppressed.

The immediatering option can now be set to no to suppress
the fake audible ringback provided when immediate=yes on FXS channels.

Upgrade Notes:

Closed Issues:

#37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on aarch64 Debian platforms
#71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to some resource
#118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when immediate=yes
#170: [improvement]: app_voicemail - add CLI commands to manipulate messages
#179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
#181: [improvement]: app_voicemail - add manager actions to display and manipulate messages
#202: [improvement]: app_queue: Add support for immediately applying queue caller priority change
#205: [new-feature]: sig_analog: Allow flash to time out to silent hold
#224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
#226: [improvement]: Apply contact_user to incoming calls
#230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
#233: [bug]: Deadlock with MixMonitorMute AMI action
#240: [new-feature]: sig_analog: Add Called Subscriber Held capability
#253: app_gosub patch appear to have broken predial handlers that utilize macros that call gosubs
#255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules registered"
#263: [bug]: download_externals doesn't always handle versions correctly
#265: [bug]: app_macro isn't locking around channel datastore access
#267: [bug]: ari: refer with display_name key in request body leads to crash
#274: [bug]: Syntax Error in SQL Code
#275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99 -Wdeclaration-after-statement'
#277: [bug]: pbx.c: Compiler error with gcc 12.2
#281: [bug]: app_dial: Infinite loop if called channel hangs up while receiving digits

asterisk - Asterisk Release 18.20.0-rc1

Published by asterisk-org-access-app[bot] about 1 year ago

The Asterisk Development Team would like to announce
release candidate 1 of asterisk-18.20.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/18.20.0-rc1
and
https://downloads.asterisk.org/pub/telephony/asterisk

This release resolves issues reported by the community
and would have not been possible without your participation.

Thank You!

Change Log for Release asterisk-18.20.0-rc1

Summary:

ari-stubs: Fix more local anchor references
ari-stubs: Fix more local anchor references
ari-stubs: Fix broken documentation anchors
res_pjsip_session: Send Session Interval too small response
.github: Update workflow-application-token-action to v2
app_dial: Fix infinite loop when sending digits.
app_voicemail: Fix for loop declarations
alembic: Fix quoting of the 100rel column
pbx.c: Fix gcc 12 compiler warning.
app_audiosocket: Fixed timeout with -1 to avoid busy loop.
download_externals: Fix a few version related issues
main/refer.c: Fix double free in refer_data_destructor + potential leak
sig_analog: Add Called Subscriber Held capability.
app_macro: Fix locking around datastore access
Revert "app_stack: Print proper exit location for PBXless channels."
.github: Use generic releaser
install_prereq: Fix dependency install on aarch64.
res_pjsip.c: Set contact_user on incoming call local Contact header
extconfig: Allow explicit DB result set ordering to be disabled.
rest-api: Run make ari-stubs
res_pjsip_header_funcs: Make prefix argument optional.
pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
manager: Tolerate stasis messages with no channel snapshot.
core/ari/pjsip: Add refer mechanism
chan_dahdi: Allow autoreoriginating after hangup.
audiohook: Unlock channel in mute if no audiohooks present.
sig_analog: Allow three-way flash to time out to silence.
res_prometheus: Do not generate broken metrics
res_pjsip: Enable TLS v1.3 if present.
func_cut: Add example to documentation.
extensions.conf.sample: Remove reference to missing context.
func_export: Use correct function argument as variable name.
app_queue: Add support for applying caller priority change immediately.
.github: Fix cherry-pick reminder issues
chan_iax2.c: Avoid crash with IAX2 switch support.
res_geolocation: Ensure required 'location_info' is present.
Adds manager actions to allow move/remove/forward individual messages in a particular mailbox folder. The forward command can be used to copy a message within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry, required to retrieve message ID's.
app_voicemail: add CLI commands for message manipulation
res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using rtp->themssrc_valid into the scope of the rtp_instance lock.
.github: Minor tweak to Asterisk Releaser
.github: Suppress cherry-pick reminder for some situations
sig_analog: Allow immediate fake ring to be suppressed.

User Notes:

sig_analog: Add Called Subscriber Held capability.

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.
res_pjsip_header_funcs: Make prefix argument optional.

The prefix argument to PJSIP_HEADERS is now
optional. If not specified, all header names will be
returned.
core/ari/pjsip: Add refer mechanism

There is a new ARI endpoint /endpoints/refer for referring
an endpoint to some URI or endpoint.
chan_dahdi: Allow autoreoriginating after hangup.

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.
sig_analog: Allow three-way flash to time out to silence.

The threewaysilenthold option now allows the three-way
dial tone to time out to silence, rather than continuing forever.
res_pjsip: Enable TLS v1.3 if present.

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.
app_queue: Add support for applying caller priority change immediately.

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.
Adds manager actions to allow move/remove/forward individual messages in a particular mailbox folder. The forward command can be used to copy a message within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry, required to retrieve message ID's.

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.
app_voicemail: add CLI commands for message manipulation

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>
sig_analog: Allow immediate fake ring to be suppressed.

The immediatering option can now be set to no to suppress
the fake audible ringback provided when immediate=yes on FXS channels.

Upgrade Notes:

Closed Issues:

#37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on aarch64 Debian platforms
#71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to some resource
#118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when immediate=yes
#170: [improvement]: app_voicemail - add CLI commands to manipulate messages
#179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
#181: [improvement]: app_voicemail - add manager actions to display and manipulate messages
#202: [improvement]: app_queue: Add support for immediately applying queue caller priority change
#205: [new-feature]: sig_analog: Allow flash to time out to silent hold
#224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
#226: [improvement]: Apply contact_user to incoming calls
#230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
#233: [bug]: Deadlock with MixMonitorMute AMI action
#240: [new-feature]: sig_analog: Add Called Subscriber Held capability
#253: app_gosub patch appear to have broken predial handlers that utilize macros that call gosubs
#255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules registered"
#263: [bug]: download_externals doesn't always handle versions correctly
#265: [bug]: app_macro isn't locking around channel datastore access
#267: [bug]: ari: refer with display_name key in request body leads to crash
#274: [bug]: Syntax Error in SQL Code
#275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99 -Wdeclaration-after-statement'
#277: [bug]: pbx.c: Compiler error with gcc 12.2
#281: [bug]: app_dial: Infinite loop if called channel hangs up while receiving digits

asterisk - Asterisk Release 18.19.0

Published by asterisk-org-access-app[bot] over 1 year ago

The Asterisk Development Team would like to announce
the release of Asterisk 18.19.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/18.19.0
and
https://downloads.asterisk.org/pub/telephony/asterisk

This release resolves issues reported by the community
and would have not been possible without your participation.

Thank You!

Change Log for Release 18.19.0

Summary:

app.h: Move declaration of ast_getdata_result before its first use
doc: Remove obsolete CHANGES-staging and UPGRADE-staging
.github: Updates for AsteriskReleaser
app_voicemail: fix imap compilation errors
res_musiconhold: avoid moh state access on unlocked chan
utils: add lock timestamps for DEBUG_THREADS
.github: Back out triggering PROpenedOrUpdated by label
.github: Move publish docs to new file CreateDocs.yml
rest-api: Updates for new documentation site
.github: Remove result check from PROpenUpdateGateTests
.github: Fix use of 'contains'
.github: Add recheck label test to additional jobs
.github: Fix recheck label typos
.github: Fix recheck label manipulation
.github: Allow PR submit checks to be re-run by label
app_voicemail_imap: Fix message count when IMAP server is unavailable
res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
res_pjsip_session: Added new function calls to avoid ABI issues.
app_queue: Add force_longest_waiting_caller option.
pjsip_transport_events.c: Use %zu printf specifier for size_t.
res_crypto.c: Gracefully handle potential key filename truncation.
configure: Remove obsolete and deprecated constructs.
res_fax_spandsp.c: Clean up a spaces/tabs issue
ast-db-manage: Synchronize revisions between comments and code.
test_statis_endpoints: Fix channel_messages test again
res_crypto.c: Avoid using the non-portable ALLPERMS macro.
tcptls: when disabling a server port, we should set the accept_fd to -1.
AMI: Add parking position parameter to Park action
test_stasis_endpoints.c: Make channel_messages more stable
build: Fix a few gcc 13 issues
.github: Rework for merge approval
ast-db-manage: Fix alembic branching error caused by #122.
app_followme: fix issue with enable_callee_prompt=no (#88)
sounds: Update download URL to use HTTPS.
configure: Makefile downloader enable follow redirects.
res_musiconhold: Add option to loop last file.
chan_dahdi: Fix Caller ID presentation for FXO ports.
AMI: Add CoreShowChannelMap action.
sig_analog: Add fuller Caller ID support.
res_stasis.c: Add new type 'sdp_label' for bridge creation.
app_queue: Preserve reason for realtime queues
.github: Fix issues with cherry-pick-reminder
indications: logging changes
.github Ignore error when adding reviewrs to PR
.github: Update field descriptions for AsteriskReleaser
callerid: Allow specifying timezone for date/time.
chan_pjsip: Allow topology/session refreshes in early media state
chan_dahdi: Fix broken hidecallerid setting.
.github: Change title of AsteriskReleaser job
asterisk.c: Fix option warning for remote console.
.github: Don't add cherry-pick reminder if it's already present
.github: Fix quoting in PROpenedOrUpdated
.github: Add cherry-pick reminder to new PRs
configure: fix test code to match gethostbyname_r prototype.
res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
res_sorcery_memory_cache.c: Fix memory leak
xml.c: Process XML Inclusions recursively.
.github: Tweak improvement issue type language.
.github: Tweak new feature language, and move feature requests elsewhere.
.github: Fix staleness check to only run on certain labels.

User Notes:

AMI: Add parking position parameter to Park action

New ParkingSpace parameter has been added to AMI action Park.
res_musiconhold: Add option to loop last file.

The loop_last option in musiconhold.conf now
allows the last file in the directory to be looped once reached.
AMI: Add CoreShowChannelMap action.

New AMI action CoreShowChannelMap has been added.
sig_analog: Add fuller Caller ID support.

Additional Caller ID properties are now supported on
incoming calls to FXS stations, namely the
redirecting reason and call qualifier.
res_stasis.c: Add new type 'sdp_label' for bridge creation.

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.
app_queue: Preserve reason for realtime queues

Make paused reason in realtime queues persist an
Asterisk restart. This was fixed for non-realtime
queues in ASTERISK_25732.

Upgrade Notes:

app_queue: Preserve reason for realtime queues
Add a new column to the queue_member table:
reason_paused VARCHAR(80) so the reason can be preserved.

Closed Issues:

#45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is insufficient/ineffective
#55: [bug]: res_sorcery_memory_cache: Memory leak when calling sorcery_memory_cache_open
#64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
#65: [bug]: heap overflow by default at startup
#66: [improvement]: Fix preserve reason of pause when Asterisk is restared for realtime queues
#73: [new-feature]: pjsip: Allow topology/session refreshes in early media state
#87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
#89: [improvement]: indications: logging changes
#91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP labels
#94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
#98: [new-feature]: callerid: Allow timezone to be specified at runtime
#100: [bug]: sig_analog: hidecallerid setting is broken
#102: [bug]: Strange warning - 'T' option is not compatible with remote console mode and has no effect.
#104: [improvement]: Add AMI action to get a list of connected channels
#108: [new-feature]: fair handling of calls in multi-queue scenarios
#110: [improvement]: utils - add lock timing information with DEBUG_THREADS
#116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
#120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
#122: [new-feature]: res_musiconhold: Add looplast option
#133: [bug]: unlock channel after moh state access
#136: [bug]: Makefile downloader does not follow redirects.
#145: [bug]: ABI issue with pjproject and pjsip_inv_session
#155: [bug]: GCC 13 is catching a few new trivial issues
#158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
#174: [bug]: app_voicemail imap compile errors
#200: [bug]: Regression: In app.h an enum is used before its declaration.

asterisk - Asterisk Release 20.4.0

Published by asterisk-org-access-app[bot] over 1 year ago

The Asterisk Development Team would like to announce
the release of Asterisk 20.4.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/20.4.0
and
https://downloads.asterisk.org/pub/telephony/asterisk

This release resolves issues reported by the community
and would have not been possible without your participation.

Thank You!

Change Log for Release 20.4.0

Summary:

app.h: Move declaration of ast_getdata_result before its first use
doc: Remove obsolete CHANGES-staging and UPGRADE-staging
.github: Updates for AsteriskReleaser
app_voicemail: fix imap compilation errors
res_musiconhold: avoid moh state access on unlocked chan
utils: add lock timestamps for DEBUG_THREADS
.github: Back out triggering PROpenedOrUpdated by label
.github: Move publish docs to new file CreateDocs.yml
rest-api: Updates for new documentation site
.github: Remove result check from PROpenUpdateGateTests
.github: Fix use of 'contains'
.github: Add recheck label test to additional jobs
.github: Fix recheck label typos
.github: Fix recheck label manipulation
.github: Allow PR submit checks to be re-run by label
app_voicemail_imap: Fix message count when IMAP server is unavailable
res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
res_pjsip_session: Added new function calls to avoid ABI issues.
app_queue: Add force_longest_waiting_caller option.
pjsip_transport_events.c: Use %zu printf specifier for size_t.
res_crypto.c: Gracefully handle potential key filename truncation.
configure: Remove obsolete and deprecated constructs.
res_fax_spandsp.c: Clean up a spaces/tabs issue
ast-db-manage: Synchronize revisions between comments and code.
test_statis_endpoints: Fix channel_messages test again
res_crypto.c: Avoid using the non-portable ALLPERMS macro.
tcptls: when disabling a server port, we should set the accept_fd to -1.
AMI: Add parking position parameter to Park action
test_stasis_endpoints.c: Make channel_messages more stable
build: Fix a few gcc 13 issues
.github: Rework for merge approval
ast-db-manage: Fix alembic branching error caused by #122.
app_followme: fix issue with enable_callee_prompt=no (#88)
sounds: Update download URL to use HTTPS.
configure: Makefile downloader enable follow redirects.
res_musiconhold: Add option to loop last file.
chan_dahdi: Fix Caller ID presentation for FXO ports.
AMI: Add CoreShowChannelMap action.
sig_analog: Add fuller Caller ID support.
res_stasis.c: Add new type 'sdp_label' for bridge creation.
app_queue: Preserve reason for realtime queues
.github: Fix issues with cherry-pick-reminder
indications: logging changes
.github Ignore error when adding reviewrs to PR
.github: Update field descriptions for AsteriskReleaser
callerid: Allow specifying timezone for date/time.
logrotate: Fix duplicate log entries.
chan_pjsip: Allow topology/session refreshes in early media state
chan_dahdi: Fix broken hidecallerid setting.
.github: Change title of AsteriskReleaser job
asterisk.c: Fix option warning for remote console.
.github: Don't add cherry-pick reminder if it's already present
.github: Fix quoting in PROpenedOrUpdated
.github: Add cherry-pick reminder to new PRs
configure: fix test code to match gethostbyname_r prototype.
res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
res_sorcery_memory_cache.c: Fix memory leak
xml.c: Process XML Inclusions recursively.
.github: Tweak improvement issue type language.
.github: Tweak new feature language, and move feature requests elsewhere.
.github: Fix staleness check to only run on certain labels.

User Notes:

AMI: Add parking position parameter to Park action

New ParkingSpace parameter has been added to AMI action Park.
res_musiconhold: Add option to loop last file.

The loop_last option in musiconhold.conf now
allows the last file in the directory to be looped once reached.
AMI: Add CoreShowChannelMap action.

New AMI action CoreShowChannelMap has been added.
sig_analog: Add fuller Caller ID support.

Additional Caller ID properties are now supported on
incoming calls to FXS stations, namely the
redirecting reason and call qualifier.
res_stasis.c: Add new type 'sdp_label' for bridge creation.

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.
app_queue: Preserve reason for realtime queues

Make paused reason in realtime queues persist an
Asterisk restart. This was fixed for non-realtime
queues in ASTERISK_25732.

Upgrade Notes:

app_queue: Preserve reason for realtime queues
Add a new column to the queue_member table:
reason_paused VARCHAR(80) so the reason can be preserved.

Closed Issues:

#45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is insufficient/ineffective
#55: [bug]: res_sorcery_memory_cache: Memory leak when calling sorcery_memory_cache_open
#64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
#65: [bug]: heap overflow by default at startup
#66: [improvement]: Fix preserve reason of pause when Asterisk is restared for realtime queues
#73: [new-feature]: pjsip: Allow topology/session refreshes in early media state
#87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
#89: [improvement]: indications: logging changes
#91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP labels
#94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
#96: [bug]: make install-logrotate causes logrotate to fail on service restart
#98: [new-feature]: callerid: Allow timezone to be specified at runtime
#100: [bug]: sig_analog: hidecallerid setting is broken
#102: [bug]: Strange warning - 'T' option is not compatible with remote console mode and has no effect.
#104: [improvement]: Add AMI action to get a list of connected channels
#108: [new-feature]: fair handling of calls in multi-queue scenarios
#110: [improvement]: utils - add lock timing information with DEBUG_THREADS
#116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
#120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
#122: [new-feature]: res_musiconhold: Add looplast option
#133: [bug]: unlock channel after moh state access
#136: [bug]: Makefile downloader does not follow redirects.
#145: [bug]: ABI issue with pjproject and pjsip_inv_session
#155: [bug]: GCC 13 is catching a few new trivial issues
#158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
#174: [bug]: app_voicemail imap compile errors
#200: [bug]: Regression: In app.h an enum is used before its declaration.

asterisk - Asterisk Release 20.4.0-rc2

Published by asterisk-org-access-app[bot] over 1 year ago

The Asterisk Development Team would like to announce
release candidate 2 of Asterisk 20.4.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/20.4.0-rc2
and
https://downloads.asterisk.org/pub/telephony/asterisk

This release resolves issues reported by the community
and would have not been possible without your participation.

Thank You!

Change Log for Release 20.4.0-rc2

Summary:

app.h: Move declaration of ast_getdata_result before its first use
doc: Remove obsolete CHANGES-staging and UPGRADE-staging

User Notes:

Upgrade Notes:

Closed Issues:

#200: [bug]: Regression: In app.h an enum is used before its declaration.

asterisk - Asterisk Release 18.19.0-rc2

Published by asterisk-org-access-app[bot] over 1 year ago

The Asterisk Development Team would like to announce
release candidate 2 of Asterisk 18.19.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/18.19.0-rc2
and
https://downloads.asterisk.org/pub/telephony/asterisk

This release resolves issues reported by the community
and would have not been possible without your participation.

Thank You!

Change Log for Release 18.19.0-rc2

Summary:

app.h: Move declaration of ast_getdata_result before its first use
doc: Remove obsolete CHANGES-staging and UPGRADE-staging

User Notes:

Upgrade Notes:

Closed Issues:

#200: [bug]: Regression: In app.h an enum is used before its declaration.

asterisk - Asterisk Release 20.4.0-rc1

Published by asterisk-org-access-app[bot] over 1 year ago

The Asterisk Development Team would like to announce
release candidate 1 of Asterisk 20.4.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/20.4.0-rc1
and
https://downloads.asterisk.org/pub/telephony/asterisk

This release resolves issues reported by the community
and would have not been possible without your participation.

Thank You!

Change Log for Release 20.4.0-rc1

Summary:

.github: Updates for AsteriskReleaser
app_voicemail: fix imap compilation errors
res_musiconhold: avoid moh state access on unlocked chan
utils: add lock timestamps for DEBUG_THREADS
.github: Back out triggering PROpenedOrUpdated by label
.github: Move publish docs to new file CreateDocs.yml
rest-api: Updates for new documentation site
.github: Remove result check from PROpenUpdateGateTests
.github: Fix use of 'contains'
.github: Add recheck label test to additional jobs
.github: Fix recheck label typos
.github: Fix recheck label manipulation
.github: Allow PR submit checks to be re-run by label
app_voicemail_imap: Fix message count when IMAP server is unavailable
res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
res_pjsip_session: Added new function calls to avoid ABI issues.
app_queue: Add force_longest_waiting_caller option.
pjsip_transport_events.c: Use %zu printf specifier for size_t.
res_crypto.c: Gracefully handle potential key filename truncation.
configure: Remove obsolete and deprecated constructs.
res_fax_spandsp.c: Clean up a spaces/tabs issue
ast-db-manage: Synchronize revisions between comments and code.
test_statis_endpoints: Fix channel_messages test again
res_crypto.c: Avoid using the non-portable ALLPERMS macro.
tcptls: when disabling a server port, we should set the accept_fd to -1.
AMI: Add parking position parameter to Park action
test_stasis_endpoints.c: Make channel_messages more stable
build: Fix a few gcc 13 issues
.github: Rework for merge approval
ast-db-manage: Fix alembic branching error caused by #122.
app_followme: fix issue with enable_callee_prompt=no (#88)
sounds: Update download URL to use HTTPS.
configure: Makefile downloader enable follow redirects.
res_musiconhold: Add option to loop last file.
chan_dahdi: Fix Caller ID presentation for FXO ports.
AMI: Add CoreShowChannelMap action.
sig_analog: Add fuller Caller ID support.
res_stasis.c: Add new type 'sdp_label' for bridge creation.
app_queue: Preserve reason for realtime queues
.github: Fix issues with cherry-pick-reminder
indications: logging changes
.github Ignore error when adding reviewrs to PR
.github: Update field descriptions for AsteriskReleaser
callerid: Allow specifying timezone for date/time.
logrotate: Fix duplicate log entries.
chan_pjsip: Allow topology/session refreshes in early media state
chan_dahdi: Fix broken hidecallerid setting.
.github: Change title of AsteriskReleaser job
asterisk.c: Fix option warning for remote console.
.github: Don't add cherry-pick reminder if it's already present
.github: Fix quoting in PROpenedOrUpdated
.github: Add cherry-pick reminder to new PRs
configure: fix test code to match gethostbyname_r prototype.
res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
res_sorcery_memory_cache.c: Fix memory leak
xml.c: Process XML Inclusions recursively.
.github: Tweak improvement issue type language.
.github: Tweak new feature language, and move feature requests elsewhere.
.github: Fix staleness check to only run on certain labels.

User Notes:

AMI: Add parking position parameter to Park action

New ParkingSpace parameter has been added to AMI action Park.
res_musiconhold: Add option to loop last file.

The loop_last option in musiconhold.conf now
allows the last file in the directory to be looped once reached.
AMI: Add CoreShowChannelMap action.

New AMI action CoreShowChannelMap has been added.
sig_analog: Add fuller Caller ID support.

Additional Caller ID properties are now supported on
incoming calls to FXS stations, namely the
redirecting reason and call qualifier.
res_stasis.c: Add new type 'sdp_label' for bridge creation.

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.
app_queue: Preserve reason for realtime queues

Make paused reason in realtime queues persist an
Asterisk restart. This was fixed for non-realtime
queues in ASTERISK_25732.
res_http_media_cache: Introduce options and customize

The res_http_media_cache module now attempts to load
configuration from the res_http_media_cache.conf file.
The following options were added:
- timeout_secs
- user_agent
- follow_location
- max_redirects
- protocols
- redirect_protocols
- dns_cache_timeout_secs
format_sln: add .slin as supported file extension

format_sln now recognizes '.slin' as a valid
file extension in addition to the existing
'.sln' and '.raw'.
bridge_builtin_features: add beep via touch variable

Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid
interval in seconds will result in a periodic beep being
played to the monitored channel upon MixMontior/Monitor
feature start.
If an interval less than 5 seconds is specified, the interval
will default to 5 seconds. If the value is set to an invalid
interval, the default of 15 seconds will be used.
app_senddtmf: Add SendFlash AMI action.

The SendFlash AMI action now allows sending
a hook flash event on a channel.
res_mixmonitor: MixMonitorMute by MixMonitor ID

It is now possible to specify the MixMonitorID when calling
the manager action: MixMonitorMute. This will allow an
individual MixMonitor instance to be muted via ID.
The MixMonitorID can be stored as a channel variable using
the 'i' MixMonitor option and is returned upon creation if
this option is used.
As part of this change, if no MixMonitorID is specified in
the manager action MixMonitorMute, Asterisk will set the mute
flag on all MixMonitor audiohooks on the channel. Previous
behavior would set the flag on the first MixMonitor audiohook
found.
pbx_dundi: Add PJSIP support.

DUNDi now supports chan_pjsip. Outgoing calls using
PJSIP require the pjsip_outgoing_endpoint option
to be set in dundi.conf.
test.c: Fix counting of tests and add 2 new tests

The "tests" attribute of the "testsuite" element in the
output XML now reflects only the tests actually requested
to be executed instead of all the tests registered.
The "failures" attribute was added to the "testsuite"
element.
Also added two new unit tests that just pass and fail
to be used for testing CI itself.
cli: increase channel column width

This change increases the display width on 'core show channels'
amd 'core show channels verbose'
For 'core show channels', the Channel name field is increased to
64 characters and the Location name field is increased to 32
characters.
For 'core show channels verbose', the Channel name field is
increased to 80 characters, the Context is increased to 24
characters and the Extension is increased to 24 characters.

Upgrade Notes:

app_queue: Preserve reason for realtime queues
Add a new column to the queue_member table:
reason_paused VARCHAR(80) so the reason can be preserved.

Closed Issues:

#45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is insufficient/ineffective
#55: [bug]: res_sorcery_memory_cache: Memory leak when calling sorcery_memory_cache_open
#64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
#65: [bug]: heap overflow by default at startup
#66: [improvement]: Fix preserve reason of pause when Asterisk is restared for realtime queues
#73: [new-feature]: pjsip: Allow topology/session refreshes in early media state
#87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
#89: [improvement]: indications: logging changes
#91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP labels
#94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
#96: [bug]: make install-logrotate causes logrotate to fail on service restart
#98: [new-feature]: callerid: Allow timezone to be specified at runtime
#100: [bug]: sig_analog: hidecallerid setting is broken
#102: [bug]: Strange warning - 'T' option is not compatible with remote console mode and has no effect.
#104: [improvement]: Add AMI action to get a list of connected channels
#108: [new-feature]: fair handling of calls in multi-queue scenarios
#110: [improvement]: utils - add lock timing information with DEBUG_THREADS
#116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
#120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
#122: [new-feature]: res_musiconhold: Add looplast option
#133: [bug]: unlock channel after moh state access
#136: [bug]: Makefile downloader does not follow redirects.
#145: [bug]: ABI issue with pjproject and pjsip_inv_session
#155: [bug]: GCC 13 is catching a few new trivial issues
#158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
#174: [bug]: app_voicemail imap compile errors

asterisk - Asterisk Release 18.19.0-rc1

Published by asterisk-org-access-app[bot] over 1 year ago

The Asterisk Development Team would like to announce
release candidate 1 of Asterisk 18.19.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/18.19.0-rc1
and
https://downloads.asterisk.org/pub/telephony/asterisk

This release resolves issues reported by the community
and would have not been possible without your participation.

Thank You!

Change Log for Release 18.19.0-rc1

Summary:

.github: Updates for AsteriskReleaser
app_voicemail: fix imap compilation errors
res_musiconhold: avoid moh state access on unlocked chan
utils: add lock timestamps for DEBUG_THREADS
.github: Back out triggering PROpenedOrUpdated by label
.github: Move publish docs to new file CreateDocs.yml
rest-api: Updates for new documentation site
.github: Remove result check from PROpenUpdateGateTests
.github: Fix use of 'contains'
.github: Add recheck label test to additional jobs
.github: Fix recheck label typos
.github: Fix recheck label manipulation
.github: Allow PR submit checks to be re-run by label
app_voicemail_imap: Fix message count when IMAP server is unavailable
res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
res_pjsip_session: Added new function calls to avoid ABI issues.
app_queue: Add force_longest_waiting_caller option.
pjsip_transport_events.c: Use %zu printf specifier for size_t.
res_crypto.c: Gracefully handle potential key filename truncation.
configure: Remove obsolete and deprecated constructs.
res_fax_spandsp.c: Clean up a spaces/tabs issue
ast-db-manage: Synchronize revisions between comments and code.
test_statis_endpoints: Fix channel_messages test again
res_crypto.c: Avoid using the non-portable ALLPERMS macro.
tcptls: when disabling a server port, we should set the accept_fd to -1.
AMI: Add parking position parameter to Park action
test_stasis_endpoints.c: Make channel_messages more stable
build: Fix a few gcc 13 issues
.github: Rework for merge approval
ast-db-manage: Fix alembic branching error caused by #122.
app_followme: fix issue with enable_callee_prompt=no (#88)
sounds: Update download URL to use HTTPS.
configure: Makefile downloader enable follow redirects.
res_musiconhold: Add option to loop last file.
chan_dahdi: Fix Caller ID presentation for FXO ports.
AMI: Add CoreShowChannelMap action.
sig_analog: Add fuller Caller ID support.
res_stasis.c: Add new type 'sdp_label' for bridge creation.
app_queue: Preserve reason for realtime queues
.github: Fix issues with cherry-pick-reminder
indications: logging changes
.github Ignore error when adding reviewrs to PR
.github: Update field descriptions for AsteriskReleaser
callerid: Allow specifying timezone for date/time.
chan_pjsip: Allow topology/session refreshes in early media state
chan_dahdi: Fix broken hidecallerid setting.
.github: Change title of AsteriskReleaser job
asterisk.c: Fix option warning for remote console.
.github: Don't add cherry-pick reminder if it's already present
.github: Fix quoting in PROpenedOrUpdated
.github: Add cherry-pick reminder to new PRs
configure: fix test code to match gethostbyname_r prototype.
res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
res_sorcery_memory_cache.c: Fix memory leak
xml.c: Process XML Inclusions recursively.
.github: Tweak improvement issue type language.
.github: Tweak new feature language, and move feature requests elsewhere.
.github: Fix staleness check to only run on certain labels.

User Notes:

AMI: Add parking position parameter to Park action

New ParkingSpace parameter has been added to AMI action Park.
res_musiconhold: Add option to loop last file.

The loop_last option in musiconhold.conf now
allows the last file in the directory to be looped once reached.
AMI: Add CoreShowChannelMap action.

New AMI action CoreShowChannelMap has been added.
sig_analog: Add fuller Caller ID support.

Additional Caller ID properties are now supported on
incoming calls to FXS stations, namely the
redirecting reason and call qualifier.
res_stasis.c: Add new type 'sdp_label' for bridge creation.

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.
app_queue: Preserve reason for realtime queues

Make paused reason in realtime queues persist an
Asterisk restart. This was fixed for non-realtime
queues in ASTERISK_25732.
res_http_media_cache: Introduce options and customize

The res_http_media_cache module now attempts to load
configuration from the res_http_media_cache.conf file.
The following options were added:
- timeout_secs
- user_agent
- follow_location
- max_redirects
- protocols
- redirect_protocols
- dns_cache_timeout_secs
format_sln: add .slin as supported file extension

format_sln now recognizes '.slin' as a valid
file extension in addition to the existing
'.sln' and '.raw'.
bridge_builtin_features: add beep via touch variable

Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid
interval in seconds will result in a periodic beep being
played to the monitored channel upon MixMontior/Monitor
feature start.
If an interval less than 5 seconds is specified, the interval
will default to 5 seconds. If the value is set to an invalid
interval, the default of 15 seconds will be used.
app_senddtmf: Add SendFlash AMI action.

The SendFlash AMI action now allows sending
a hook flash event on a channel.
res_mixmonitor: MixMonitorMute by MixMonitor ID

It is now possible to specify the MixMonitorID when calling
the manager action: MixMonitorMute. This will allow an
individual MixMonitor instance to be muted via ID.
The MixMonitorID can be stored as a channel variable using
the 'i' MixMonitor option and is returned upon creation if
this option is used.
As part of this change, if no MixMonitorID is specified in
the manager action MixMonitorMute, Asterisk will set the mute
flag on all MixMonitor audiohooks on the channel. Previous
behavior would set the flag on the first MixMonitor audiohook
found.
pbx_dundi: Add PJSIP support.

DUNDi now supports chan_pjsip. Outgoing calls using
PJSIP require the pjsip_outgoing_endpoint option
to be set in dundi.conf.
test.c: Fix counting of tests and add 2 new tests

The "tests" attribute of the "testsuite" element in the
output XML now reflects only the tests actually requested
to be executed instead of all the tests registered.
The "failures" attribute was added to the "testsuite"
element.
Also added two new unit tests that just pass and fail
to be used for testing CI itself.
cli: increase channel column width

This change increases the display width on 'core show channels'
amd 'core show channels verbose'
For 'core show channels', the Channel name field is increased to
64 characters and the Location name field is increased to 32
characters.
For 'core show channels verbose', the Channel name field is
increased to 80 characters, the Context is increased to 24
characters and the Extension is increased to 24 characters.

Upgrade Notes:

app_queue: Preserve reason for realtime queues
Add a new column to the queue_member table:
reason_paused VARCHAR(80) so the reason can be preserved.

Closed Issues:

#45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is insufficient/ineffective
#55: [bug]: res_sorcery_memory_cache: Memory leak when calling sorcery_memory_cache_open
#64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
#65: [bug]: heap overflow by default at startup
#66: [improvement]: Fix preserve reason of pause when Asterisk is restared for realtime queues
#73: [new-feature]: pjsip: Allow topology/session refreshes in early media state
#87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
#89: [improvement]: indications: logging changes
#91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP labels
#94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
#98: [new-feature]: callerid: Allow timezone to be specified at runtime
#100: [bug]: sig_analog: hidecallerid setting is broken
#102: [bug]: Strange warning - 'T' option is not compatible with remote console mode and has no effect.
#104: [improvement]: Add AMI action to get a list of connected channels
#108: [new-feature]: fair handling of calls in multi-queue scenarios
#110: [improvement]: utils - add lock timing information with DEBUG_THREADS
#116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
#120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
#122: [new-feature]: res_musiconhold: Add looplast option
#133: [bug]: unlock channel after moh state access
#136: [bug]: Makefile downloader does not follow redirects.
#145: [bug]: ABI issue with pjproject and pjsip_inv_session
#155: [bug]: GCC 13 is catching a few new trivial issues
#158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
#174: [bug]: app_voicemail imap compile errors

asterisk - Asterisk Release 20.3.1

Published by asterisk-org-access-app[bot] over 1 year ago

The Asterisk Development Team would like to announce security release
Asterisk 20.3.1.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/20.3.1
and
https://downloads.asterisk.org/pub/telephony/asterisk

The following security advisories were resolved in this release:
https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm

Change Log for Release 20.3.1

Summary:

apply_patches: Use globbing instead of file/sort.
apply_patches: Sort patch list before applying
pjsip: Upgrade bundled version to pjproject 2.13.1

User Notes:

res_http_media_cache: Introduce options and customize

The res_http_media_cache module now attempts to load
configuration from the res_http_media_cache.conf file.
The following options were added:
- timeout_secs
- user_agent
- follow_location
- max_redirects
- protocols
- redirect_protocols
- dns_cache_timeout_secs
format_sln: add .slin as supported file extension

format_sln now recognizes '.slin' as a valid
file extension in addition to the existing
'.sln' and '.raw'.
bridge_builtin_features: add beep via touch variable

Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid
interval in seconds will result in a periodic beep being
played to the monitored channel upon MixMontior/Monitor
feature start.
If an interval less than 5 seconds is specified, the interval
will default to 5 seconds. If the value is set to an invalid
interval, the default of 15 seconds will be used.
app_senddtmf: Add SendFlash AMI action.

The SendFlash AMI action now allows sending
a hook flash event on a channel.
res_mixmonitor: MixMonitorMute by MixMonitor ID

It is now possible to specify the MixMonitorID when calling
the manager action: MixMonitorMute. This will allow an
individual MixMonitor instance to be muted via ID.
The MixMonitorID can be stored as a channel variable using
the 'i' MixMonitor option and is returned upon creation if
this option is used.
As part of this change, if no MixMonitorID is specified in
the manager action MixMonitorMute, Asterisk will set the mute
flag on all MixMonitor audiohooks on the channel. Previous
behavior would set the flag on the first MixMonitor audiohook
found.
pbx_dundi: Add PJSIP support.

DUNDi now supports chan_pjsip. Outgoing calls using
PJSIP require the pjsip_outgoing_endpoint option
to be set in dundi.conf.
test.c: Fix counting of tests and add 2 new tests

The "tests" attribute of the "testsuite" element in the
output XML now reflects only the tests actually requested
to be executed instead of all the tests registered.
The "failures" attribute was added to the "testsuite"
element.
Also added two new unit tests that just pass and fail
to be used for testing CI itself.
cli: increase channel column width

This change increases the display width on 'core show channels'
amd 'core show channels verbose'
For 'core show channels', the Channel name field is increased to
64 characters and the Location name field is increased to 32
characters.
For 'core show channels verbose', the Channel name field is
increased to 80 characters, the Context is increased to 24
characters and the Extension is increased to 24 characters.

Upgrade Notes:

Closed Issues:

#193: [bug]: third-party/apply-patches doesn't sort the patch file list before applying

asterisk - Asterisk Release certified-18.9-cert5

Published by asterisk-org-access-app[bot] over 1 year ago

The Asterisk Development Team would like to announce security release
Certified Asterisk 18.9-cert5.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/certified-18.9-cert5
and
https://downloads.asterisk.org/pub/telephony/certified-asterisk

The following security advisories were resolved in this release:
https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm

Change Log for Release certified-18.9-cert5

Summary:

apply_patches: Use globbing instead of file/sort.
apply_patches: Sort patch list before applying
bundled_pjproject: Backport security fixes from pjproject 2.13.1
.github: Updates for AsteriskReleaser
res_musiconhold: avoid moh state access on unlocked chan
utils: add lock timestamps for DEBUG_THREADS
.github: Back out triggering PROpenedOrUpdated by label
.github: Move publish docs to new file CreateDocs.yml
.github: Remove result check from PROpenUpdateGateTests
.github: Fix use of 'contains'
.github: Add recheck label test to additional jobs
.github: Fix recheck label typos
.github: Fix recheck label manipulation
.github: Allow PR submit checks to be re-run by label
res_pjsip_session: Added new function calls to avoid ABI issues.
test_statis_endpoints: Fix channel_messages test again
test_stasis_endpoints.c: Make channel_messages more stable
build: Fix a few gcc 13 issues
.github: Rework for merge approval
AMI: Add CoreShowChannelMap action.
.github: Fix issues with cherry-pick-reminder
indications: logging changes
.github Ignore error when adding reviewrs to PR
.github: Update field descriptions for AsteriskReleaser
.github: Change title of AsteriskReleaser job
.github: Don't add cherry-pick reminder if it's already present
.github: Fix quoting in PROpenedOrUpdated
.github: Add cherry-pick reminder to new PRs
core: Cleanup gerrit and JIRA references. (#40) (#61)
.github: Tweak improvement issue type language.
.github: Tweak new feature language, and move feature requests elsewhere.
.github: Fix staleness check to only run on certain labels.
.github: Add AsteriskReleaser
cel: add local optimization begin event
.github: Fix CherryPickTest to only run when it should
.github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
.github: Remove separate set labels step from new PR
.github: Refactor CP progress and add new PR test progress
.github: Add cherry-pick test progress labels
.github: Update issue templates
.github: Remove unnecessary parameter in CherryPickTest
Initial GitHub PRs
Initial GitHub Issue Templates
test.c: Fix counting of tests and add 2 new tests
res_mixmonitor: MixMonitorMute by MixMonitor ID
format_sln: add .slin as supported file extension
bridge_builtin_features: add beep via touch variable
cli: increase channel column width
app_senddtmf: Add option to answer target channel.
app_directory: Add a 'skip call' option.
app_read: Add an option to return terminator on empty digits.
app_directory: add ability to specify configuration file

User Notes:

AMI: Add CoreShowChannelMap action.

New AMI action CoreShowChannelMap has been added.
cel: add local optimization begin event

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.
app_read: Add an option to return terminator on empty digits.

A new option 'e' has been added to allow Read() to return the
terminator as the dialed digits in the case where only the terminator
is entered.
format_sln: add .slin as supported file extension

format_sln now recognizes '.slin' as a valid
file extension in addition to the existing
'.sln' and '.raw'.
bridge_builtin_features: add beep via touch variable

Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid
interval in seconds will result in a periodic beep being
played to the monitored channel upon MixMontior/Monitor
feature start.
If an interval less than 5 seconds is specified, the interval
will default to 5 seconds. If the value is set to an invalid
interval, the default of 15 seconds will be used.
app_directory: Add a 'skip call' option.

A new option 's' has been added to the Directory() application that
will skip calling the extension and instead set the extension as
DIRECTORY_EXTEN channel variable.
res_mixmonitor: MixMonitorMute by MixMonitor ID

It is now possible to specify the MixMonitorID when calling
the manager action: MixMonitorMute. This will allow an
individual MixMonitor instance to be muted via ID.
The MixMonitorID can be stored as a channel variable using
the 'i' MixMonitor option and is returned upon creation if
this option is used.
As part of this change, if no MixMonitorID is specified in
the manager action MixMonitorMute, Asterisk will set the mute
flag on all MixMonitor audiohooks on the channel. Previous
behavior would set the flag on the first MixMonitor audiohook
found.
app_senddtmf: Add option to answer target channel.

A new option has been added to SendDTMF() which will answer the
specified channel if it is not already up. If no channel is specified,
the current channel will be answered instead.
test.c: Fix counting of tests and add 2 new tests

The "tests" attribute of the "testsuite" element in the
output XML now reflects only the tests actually requested
to be executed instead of all the tests registered.
The "failures" attribute was added to the "testsuite"
element.
Also added two new unit tests that just pass and fail
to be used for testing CI itself.
cli: increase channel column width

This change increases the display width on 'core show channels'
amd 'core show channels verbose'
For 'core show channels', the Channel name field is increased to
64 characters and the Location name field is increased to 32
characters.
For 'core show channels verbose', the Channel name field is
increased to 80 characters, the Context is increased to 24
characters and the Extension is increased to 24 characters.

Upgrade Notes:

cel: add local optimization begin event
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.

Closed Issues:

#39: [Bug]: Remove .gitreview from repository.
#52: [improvement]: Add local optimization begin cel event
#89: [improvement]: indications: logging changes
#104: [improvement]: Add AMI action to get a list of connected channels
#110: [improvement]: utils - add lock timing information with DEBUG_THREADS
#133: [bug]: unlock channel after moh state access
#145: [bug]: ABI issue with pjproject and pjsip_inv_session
#155: [bug]: GCC 13 is catching a few new trivial issues
#158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
#188: [improvement]: pjsip: Upgrade bundled version to pjproject 2.13.1 #187
#193: [bug]: third-party/apply-patches doesn't sort the patch file list before applying

asterisk - Asterisk Release 19.8.1

Published by asterisk-org-access-app[bot] over 1 year ago

The Asterisk Development Team would like to announce security release
Asterisk 19.8.1.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/19.8.1
and
https://downloads.asterisk.org/pub/telephony/asterisk

The following security advisories were resolved in this release:
https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm

Change Log for Release 19.8.1

Summary:

apply_patches: Use globbing instead of file/sort.
bundled_pjproject: Backport 2 SSL patches from upstream
bundled_pjproject: Backport security fixes from pjproject 2.13.1
apply_patches: Sort patch list before applying

User Notes:

Upgrade Notes:

Closed Issues:

#188: [improvement]: pjsip: Upgrade bundled version to pjproject 2.13.1 #187
#193: [bug]: third-party/apply-patches doesn't sort the patch file list before applying
#194: [bug]: Segfault/double-free in bundled pjproject using TLS transport

asterisk - Asterisk Release 18.18.1

Published by asterisk-org-access-app[bot] over 1 year ago

The Asterisk Development Team would like to announce security release
Asterisk 18.18.1.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/18.18.1
and
https://downloads.asterisk.org/pub/telephony/asterisk

The following security advisories were resolved in this release:
https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm

Change Log for Release 18.18.1

Summary:

apply_patches: Use globbing instead of file/sort.
apply_patches: Sort patch list before applying
pjsip: Upgrade bundled version to pjproject 2.13.1

User Notes:

res_http_media_cache: Introduce options and customize

The res_http_media_cache module now attempts to load
configuration from the res_http_media_cache.conf file.
The following options were added:
- timeout_secs
- user_agent
- follow_location
- max_redirects
- protocols
- redirect_protocols
- dns_cache_timeout_secs
format_sln: add .slin as supported file extension

format_sln now recognizes '.slin' as a valid
file extension in addition to the existing
'.sln' and '.raw'.
bridge_builtin_features: add beep via touch variable

Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid
interval in seconds will result in a periodic beep being
played to the monitored channel upon MixMontior/Monitor
feature start.
If an interval less than 5 seconds is specified, the interval
will default to 5 seconds. If the value is set to an invalid
interval, the default of 15 seconds will be used.
app_senddtmf: Add SendFlash AMI action.

The SendFlash AMI action now allows sending
a hook flash event on a channel.
res_mixmonitor: MixMonitorMute by MixMonitor ID

It is now possible to specify the MixMonitorID when calling
the manager action: MixMonitorMute. This will allow an
individual MixMonitor instance to be muted via ID.
The MixMonitorID can be stored as a channel variable using
the 'i' MixMonitor option and is returned upon creation if
this option is used.
As part of this change, if no MixMonitorID is specified in
the manager action MixMonitorMute, Asterisk will set the mute
flag on all MixMonitor audiohooks on the channel. Previous
behavior would set the flag on the first MixMonitor audiohook
found.
pbx_dundi: Add PJSIP support.

DUNDi now supports chan_pjsip. Outgoing calls using
PJSIP require the pjsip_outgoing_endpoint option
to be set in dundi.conf.
test.c: Fix counting of tests and add 2 new tests

The "tests" attribute of the "testsuite" element in the
output XML now reflects only the tests actually requested
to be executed instead of all the tests registered.
The "failures" attribute was added to the "testsuite"
element.
Also added two new unit tests that just pass and fail
to be used for testing CI itself.
cli: increase channel column width

This change increases the display width on 'core show channels'
amd 'core show channels verbose'
For 'core show channels', the Channel name field is increased to
64 characters and the Location name field is increased to 32
characters.
For 'core show channels verbose', the Channel name field is
increased to 80 characters, the Context is increased to 24
characters and the Extension is increased to 24 characters.

Upgrade Notes:

Closed Issues:

#193: [bug]: third-party/apply-patches doesn't sort the patch file list before applying

asterisk - Asterisk Release 16.30.1

Published by asterisk-org-access-app[bot] over 1 year ago

The Asterisk Development Team would like to announce security release
Asterisk 16.30.1.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/16.30.1
and
https://downloads.asterisk.org/pub/telephony/asterisk

The following security advisories were resolved in this release:
https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm

Change Log for Release 16.30.1

Summary:

apply_patches: Use globbing instead of file/sort.
bundled_pjproject: Backport 2 SSL patches from upstream
bundled_pjproject: Backport security fixes from pjproject 2.13.1
apply_patches: Sort patch list before applying

User Notes:

Upgrade Notes:

Closed Issues:

#188: [improvement]: pjsip: Upgrade bundled version to pjproject 2.13.1 #187
#193: [bug]: third-party/apply-patches doesn't sort the patch file list before applying
#194: [bug]: Segfault/double-free in bundled pjproject using TLS transport

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asterisk

Change Log for Release asterisk-20.5.1

Links:

Summary:

User Notes:

Upgrade Notes:

Closed Issues:

Change Log for Release asterisk-18.20.1

Links:

Summary:

User Notes:

Upgrade Notes:

Closed Issues:

Change Log for Release asterisk-21.0.0

Links:

Summary:

User Notes:

sig_analog: Add Called Subscriber Held capability.

res_pjsip_header_funcs: Make prefix argument optional.

http.c: Minor simplification to HTTP status output.

Upgrade Notes:

utils.h: Deprecate ast_gethostbyname(). (#79)

app_sla: Migrate SLA applications out of app_meetme.

users.conf: Deprecate users.conf configuration.

res_monitor: Remove deprecated module.

app_osplookup: Remove deprecated module.

app_cdr: Remove deprecated application and option.

app_macro: Remove deprecated module.

translate.c: Prefer better codecs upon translate ties.

chan_sip: Remove deprecated module.

chan_alsa: Remove deprecated module.

pbx_builtins: Remove deprecated and defunct functionality.

chan_mgcp: Remove deprecated module.

chan_skinny: Remove deprecated module.

Closed Issues:

Change Log for Release asterisk-20.5.0

Links:

Summary:

User Notes:

sig_analog: Add Called Subscriber Held capability.

res_pjsip_header_funcs: Make prefix argument optional.

core/ari/pjsip: Add refer mechanism

chan_dahdi: Allow autoreoriginating after hangup.

sig_analog: Allow three-way flash to time out to silence.

res_pjsip: Enable TLS v1.3 if present.

app_queue: Add support for applying caller priority change immediately.

Adds manager actions to allow move/remove/forward individual messages in a particular mailbox folder. The forward command can be used to copy a message within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry, required to retrieve message ID's.

app_voicemail: add CLI commands for message manipulation

sig_analog: Allow immediate fake ring to be suppressed.

Upgrade Notes:

Closed Issues:

Change Log for Release asterisk-18.20.0

Links:

Summary:

User Notes:

sig_analog: Add Called Subscriber Held capability.

res_pjsip_header_funcs: Make prefix argument optional.

core/ari/pjsip: Add refer mechanism

chan_dahdi: Allow autoreoriginating after hangup.

sig_analog: Allow three-way flash to time out to silence.

res_pjsip: Enable TLS v1.3 if present.

app_queue: Add support for applying caller priority change immediately.

Adds manager actions to allow move/remove/forward individual messages in a particular mailbox folder. The forward command can be used to copy a message within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry, required to retrieve message ID's.

app_voicemail: add CLI commands for message manipulation

sig_analog: Allow immediate fake ring to be suppressed.

Upgrade Notes:

Closed Issues:

Change Log for Release asterisk-21.0.0-rc1

Links:

Summary:

User Notes:

sig_analog: Add Called Subscriber Held capability.

res_pjsip_header_funcs: Make prefix argument optional.

http.c: Minor simplification to HTTP status output.

Upgrade Notes:

utils.h: Deprecate ast_gethostbyname(). (#79)

app_sla: Migrate SLA applications out of app_meetme.

users.conf: Deprecate users.conf configuration.

app_osplookup: Remove deprecated module.

res_monitor: Remove deprecated module.

utils.h: Deprecate `ast_gethostbyname()`. (#79)

utils.h: Deprecate `ast_gethostbyname()`. (#79)

utils.h: Deprecate `ast_gethostbyname()`. (#79)