Bot releases are visible (Hide)

asterisk - Asterisk Release 21.2.0-rc1

Published by asterisk-org-access-app[bot] 8 months ago

The Asterisk Development Team would like to announce
release candidate 1 of asterisk-21.2.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/21.2.0-rc1
and
https://downloads.asterisk.org/pub/telephony/asterisk

Repository: https://github.com/asterisk/asterisk
Tag: 21.2.0-rc1

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

Thank You!

Change Log for Release asterisk-21.2.0-rc1

Summary:

app_dial: Add dial time for progress/ringing.
app_voicemail: Properly reinitialize config after unit tests.
app_queue.c : fix "queue add member" usage string
app_voicemail: Allow preventing mark messages as urgent.
res_pjsip: Use consistent type for boolean columns.
attestation_config.c: Use ast_free instead of ast_std_free
Makefile: Add stir_shaken/cache to directories created on install
Stir/Shaken Refactor
translate.c: implement new direct comp table mode
README.md: Removed outdated link
strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
res_rtp_asterisk.c: Correct coefficient in MOS calculation.
dsp.c: Fix and improve potentially inaccurate log message.
pjsip show channelstats: Prevent possible segfault when faxing
Reduce startup/shutdown verbose logging
configure: Rerun bootstrap on modern platform.
Upgrade bundled pjproject to 2.14.
res_pjsip_outbound_registration.c: Add User-Agent header override
app_speech_utils.c: Allow partial speech results.
utils: Make behavior of ast_strsep* match strsep.
app_chanspy: Add 'D' option for dual-channel audio
app_if: Fix next priority calculation.
res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
BuildSystem: Bump autotools versions on OpenBSD.
main/utils: Simplify the FreeBSD ast_get_tid() handling
res_pjsip_session.c: Correctly format SDP connection addresses.
rtp_engine.c: Correct sample rate typo for L16/44100.
manager.c: Fix erroneous reloads in UpdateConfig.
res_calendar_icalendar: Print iCalendar error on parsing failure.
app_confbridge: Don't emit warnings on valid configurations.
app_voicemail_odbc: remove macrocontext from voicemail_messages table
chan_dahdi: Allow MWI to be manually toggled on channels.
chan_rtp.c: MulticastRTP missing refcount without codec option
chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
func_frame_trace: Add CLI command to dump frame queue.

User Notes:

app_dial: Add dial time for progress/ringing.

The timeout argument to Dial now allows
specifying the maximum amount of time to dial if
early media is not received.
app_voicemail: Allow preventing mark messages as urgent.

The leaveurgent mailbox option can now be used to
control whether callers may leave messages marked as 'Urgent'.
Stir/Shaken Refactor

Asterisk's stir-shaken feature has been refactored to
correct interoperability, RFC compliance, and performance issues.
See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more
information.
Upgrade bundled pjproject to 2.14.

Bundled pjproject has been upgraded to 2.14. For more
information on what all is included in this change, check out the
pjproject Github page: https://github.com/pjsip/pjproject/releases
res_pjsip_outbound_registration.c: Add User-Agent header override

PJSIP outbound registrations now support a per-registration
User-Agent header
app_speech_utils.c: Allow partial speech results.

The SpeechBackground dialplan application now supports a 'p'
option that will return partial results from speech engines that
provide them when a timeout occurs.
app_chanspy: Add 'D' option for dual-channel audio

The ChanSpy application now accepts the 'D' option which
will interleave the spied audio within the outgoing frames. The
purpose of this is to allow the audio to be read as a Dual channel
stream with separate incoming and outgoing audio. Setting both the
'o' option and the 'D' option and results in the 'D' option being
ignored.
app_voicemail_odbc: remove macrocontext from voicemail_messages table

The fix requires removing the macrocontext column from the
voicemail_messages table in the voicemail database via alembic upgrade.
chan_dahdi: Allow MWI to be manually toggled on channels.

The 'dahdi set mwi' now allows MWI on channels
to be manually toggled if needed for troubleshooting.
Resolves: #440

Upgrade Notes:

Stir/Shaken Refactor

The stir-shaken refactor is a breaking change but since
it's not working now we don't think it matters. The
stir_shaken.conf file has changed significantly which means that
existing ones WILL need to be changed. The stir_shaken.conf.sample
file in configs/samples/ has quite a bit more information. This is
also an ABI breaking change since some of the existing objects
needed to be changed or removed, and new ones added. Additionally,
if res_stir_shaken is enabled in menuselect, you'll need to either
have the development package for libjwt v1.15.3 installed or use
the --with-libjwt-bundled option with ./configure.
app_voicemail_odbc: remove macrocontext from voicemail_messages table

The fix requires that the voicemail database be upgraded via
alembic. Upgrading to the latest voicemail database via alembic will
remove the macrocontext column from the voicemail_messages table.

Closed Issues:

#46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
#351: [improvement]: Refactor res_stir_shaken to use libjwt
#406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
#440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
#492: [improvement]: res_calendar_icalendar: Print icalendar error if available on parsing failure
#515: [improvement]: Implement option to override User-Agent-Header on a per-registration basis
#527: [bug]: app_voicemail_odbc no longer working after removal of macrocontext.
#529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed assertion bad magic number 0x0 for object" after ~30 calls
#533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of channel frame queue
#551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
#560: [bug]: EndIf() causes next priority to be skipped
#565: [bug]: Application Read() returns immediately
#569: [improvement]: Add option to interleave input and output frames on spied channel
#572: [improvement]: Copy partial speech results when Asterisk is ready to move on but the speech backend is not
#582: [improvement]: Reduce unneeded logging during startup and shutdown
#586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in older gcc versions
#588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not received
#592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault when a fax session is active
#595: [improvement]: dsp.c: Fix and improve confusing warning message.
#597: [bug]: wrong MOS calculation
#601: [new-feature]: translate.c: implement new direct comp table mode (PR #585)
#619: [new-feature]: app_voicemail: Allow preventing callers from marking messages as urgent
#629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
#634: [bug]: make install doesn't create the stir_shaken cache directory
#636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function

asterisk - Asterisk Release 20.7.0-rc1

Published by asterisk-org-access-app[bot] 8 months ago

The Asterisk Development Team would like to announce
release candidate 1 of asterisk-20.7.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/20.7.0-rc1
and
https://downloads.asterisk.org/pub/telephony/asterisk

Repository: https://github.com/asterisk/asterisk
Tag: 20.7.0-rc1

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

Thank You!

Change Log for Release asterisk-20.7.0-rc1

Summary:

app_dial: Add dial time for progress/ringing.
app_voicemail: Properly reinitialize config after unit tests.
app_queue.c : fix "queue add member" usage string
app_voicemail: Allow preventing mark messages as urgent.
res_pjsip: Use consistent type for boolean columns.
attestation_config.c: Use ast_free instead of ast_std_free
Makefile: Add stir_shaken/cache to directories created on install
Stir/Shaken Refactor
alembic: Synchronize alembic heads between supported branches.
translate.c: implement new direct comp table mode
README.md: Removed outdated link
strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
res_rtp_asterisk.c: Correct coefficient in MOS calculation.
dsp.c: Fix and improve potentially inaccurate log message.
pjsip show channelstats: Prevent possible segfault when faxing
Reduce startup/shutdown verbose logging
configure: Rerun bootstrap on modern platform.
Upgrade bundled pjproject to 2.14.
app_speech_utils.c: Allow partial speech results.
utils: Make behavior of ast_strsep* match strsep.
app_chanspy: Add 'D' option for dual-channel audio
app_if: Fix next priority calculation.
res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
BuildSystem: Bump autotools versions on OpenBSD.
main/utils: Simplify the FreeBSD ast_get_tid() handling
res_pjsip_session.c: Correctly format SDP connection addresses.
rtp_engine.c: Correct sample rate typo for L16/44100.
manager.c: Fix erroneous reloads in UpdateConfig.
res_calendar_icalendar: Print iCalendar error on parsing failure.
app_confbridge: Don't emit warnings on valid configurations.
app_voicemail: add NoOp alembic script to maintain sync
chan_dahdi: Allow MWI to be manually toggled on channels.
chan_rtp.c: MulticastRTP missing refcount without codec option
chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
func_frame_trace: Add CLI command to dump frame queue.

User Notes:

app_dial: Add dial time for progress/ringing.

The timeout argument to Dial now allows
specifying the maximum amount of time to dial if
early media is not received.
app_voicemail: Allow preventing mark messages as urgent.

The leaveurgent mailbox option can now be used to
control whether callers may leave messages marked as 'Urgent'.
Stir/Shaken Refactor

Asterisk's stir-shaken feature has been refactored to
correct interoperability, RFC compliance, and performance issues.
See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more
information.
Upgrade bundled pjproject to 2.14.

Bundled pjproject has been upgraded to 2.14. For more
information on what all is included in this change, check out the
pjproject Github page: https://github.com/pjsip/pjproject/releases
app_speech_utils.c: Allow partial speech results.

The SpeechBackground dialplan application now supports a 'p'
option that will return partial results from speech engines that
provide them when a timeout occurs.
app_chanspy: Add 'D' option for dual-channel audio

The ChanSpy application now accepts the 'D' option which
will interleave the spied audio within the outgoing frames. The
purpose of this is to allow the audio to be read as a Dual channel
stream with separate incoming and outgoing audio. Setting both the
'o' option and the 'D' option and results in the 'D' option being
ignored.
chan_dahdi: Allow MWI to be manually toggled on channels.

The 'dahdi set mwi' now allows MWI on channels
to be manually toggled if needed for troubleshooting.
Resolves: #440

Upgrade Notes:

Stir/Shaken Refactor
The stir-shaken refactor is a breaking change but since
it's not working now we don't think it matters. The
stir_shaken.conf file has changed significantly which means that
existing ones WILL need to be changed. The stir_shaken.conf.sample
file in configs/samples/ has quite a bit more information. This is
also an ABI breaking change since some of the existing objects
needed to be changed or removed, and new ones added. Additionally,
if res_stir_shaken is enabled in menuselect, you'll need to either
have the development package for libjwt v1.15.3 installed or use
the --with-libjwt-bundled option with ./configure.

Closed Issues:

#46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
#351: [improvement]: Refactor res_stir_shaken to use libjwt
#406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
#440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
#492: [improvement]: res_calendar_icalendar: Print icalendar error if available on parsing failure
#527: [bug]: app_voicemail_odbc no longer working after removal of macrocontext.
#529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed assertion bad magic number 0x0 for object" after ~30 calls
#533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of channel frame queue
#551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
#560: [bug]: EndIf() causes next priority to be skipped
#565: [bug]: Application Read() returns immediately
#569: [improvement]: Add option to interleave input and output frames on spied channel
#572: [improvement]: Copy partial speech results when Asterisk is ready to move on but the speech backend is not
#582: [improvement]: Reduce unneeded logging during startup and shutdown
#586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in older gcc versions
#588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not received
#592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault when a fax session is active
#595: [improvement]: dsp.c: Fix and improve confusing warning message.
#597: [bug]: wrong MOS calculation
#601: [new-feature]: translate.c: implement new direct comp table mode (PR #585)
#619: [new-feature]: app_voicemail: Allow preventing callers from marking messages as urgent
#629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
#634: [bug]: make install doesn't create the stir_shaken cache directory
#636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function

asterisk - Asterisk Release 18.22.0-rc1

Published by asterisk-org-access-app[bot] 8 months ago

The Asterisk Development Team would like to announce
release candidate 1 of asterisk-18.22.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/18.22.0-rc1
and
https://downloads.asterisk.org/pub/telephony/asterisk

Repository: https://github.com/asterisk/asterisk
Tag: 18.22.0-rc1

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

Thank You!

Change Log for Release asterisk-18.22.0-rc1

Summary:

app_dial: Add dial time for progress/ringing.
app_voicemail: Properly reinitialize config after unit tests.
app_queue.c : fix "queue add member" usage string
app_voicemail: Allow preventing mark messages as urgent.
res_pjsip: Use consistent type for boolean columns.
attestation_config.c: Use ast_free instead of ast_std_free
Makefile: Add stir_shaken/cache to directories created on install
Stir/Shaken Refactor
alembic: Synchronize alembic heads between supported branches.
translate.c: implement new direct comp table mode
README.md: Removed outdated link
strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
res_rtp_asterisk.c: Correct coefficient in MOS calculation.
dsp.c: Fix and improve potentially inaccurate log message.
pjsip show channelstats: Prevent possible segfault when faxing
Reduce startup/shutdown verbose logging
configure: Rerun bootstrap on modern platform.
Upgrade bundled pjproject to 2.14.
app_speech_utils.c: Allow partial speech results.
utils: Make behavior of ast_strsep* match strsep.
app_chanspy: Add 'D' option for dual-channel audio
app_if: Fix next priority calculation.
res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
BuildSystem: Bump autotools versions on OpenBSD.
main/utils: Simplify the FreeBSD ast_get_tid() handling
res_pjsip_session.c: Correctly format SDP connection addresses.
rtp_engine.c: Correct sample rate typo for L16/44100.
manager.c: Fix erroneous reloads in UpdateConfig.
res_calendar_icalendar: Print iCalendar error on parsing failure.
app_confbridge: Don't emit warnings on valid configurations.
app_voicemail: add NoOp alembic script to maintain sync
chan_dahdi: Allow MWI to be manually toggled on channels.
chan_rtp.c: MulticastRTP missing refcount without codec option
chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
func_frame_trace: Add CLI command to dump frame queue.

User Notes:

app_dial: Add dial time for progress/ringing.

The timeout argument to Dial now allows
specifying the maximum amount of time to dial if
early media is not received.
app_voicemail: Allow preventing mark messages as urgent.

The leaveurgent mailbox option can now be used to
control whether callers may leave messages marked as 'Urgent'.
Stir/Shaken Refactor

Asterisk's stir-shaken feature has been refactored to
correct interoperability, RFC compliance, and performance issues.
See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more
information.
Upgrade bundled pjproject to 2.14.

Bundled pjproject has been upgraded to 2.14. For more
information on what all is included in this change, check out the
pjproject Github page: https://github.com/pjsip/pjproject/releases
app_speech_utils.c: Allow partial speech results.

The SpeechBackground dialplan application now supports a 'p'
option that will return partial results from speech engines that
provide them when a timeout occurs.
app_chanspy: Add 'D' option for dual-channel audio

The ChanSpy application now accepts the 'D' option which
will interleave the spied audio within the outgoing frames. The
purpose of this is to allow the audio to be read as a Dual channel
stream with separate incoming and outgoing audio. Setting both the
'o' option and the 'D' option and results in the 'D' option being
ignored.
chan_dahdi: Allow MWI to be manually toggled on channels.

The 'dahdi set mwi' now allows MWI on channels
to be manually toggled if needed for troubleshooting.
Resolves: #440

Upgrade Notes:

Stir/Shaken Refactor
The stir-shaken refactor is a breaking change but since
it's not working now we don't think it matters. The
stir_shaken.conf file has changed significantly which means that
existing ones WILL need to be changed. The stir_shaken.conf.sample
file in configs/samples/ has quite a bit more information. This is
also an ABI breaking change since some of the existing objects
needed to be changed or removed, and new ones added. Additionally,
if res_stir_shaken is enabled in menuselect, you'll need to either
have the development package for libjwt v1.15.3 installed or use
the --with-libjwt-bundled option with ./configure.

Closed Issues:

#46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
#351: [improvement]: Refactor res_stir_shaken to use libjwt
#406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
#440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
#492: [improvement]: res_calendar_icalendar: Print icalendar error if available on parsing failure
#527: [bug]: app_voicemail_odbc no longer working after removal of macrocontext.
#529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed assertion bad magic number 0x0 for object" after ~30 calls
#533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of channel frame queue
#551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
#560: [bug]: EndIf() causes next priority to be skipped
#565: [bug]: Application Read() returns immediately
#569: [improvement]: Add option to interleave input and output frames on spied channel
#572: [improvement]: Copy partial speech results when Asterisk is ready to move on but the speech backend is not
#582: [improvement]: Reduce unneeded logging during startup and shutdown
#586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in older gcc versions
#588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not received
#592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault when a fax session is active
#595: [improvement]: dsp.c: Fix and improve confusing warning message.
#597: [bug]: wrong MOS calculation
#601: [new-feature]: translate.c: implement new direct comp table mode (PR #585)
#619: [new-feature]: app_voicemail: Allow preventing callers from marking messages as urgent
#629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
#634: [bug]: make install doesn't create the stir_shaken cache directory
#636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function

asterisk - Asterisk Release certified-18.9-cert8-rc2

Published by asterisk-org-access-app[bot] 8 months ago

The Asterisk Development Team would like to announce
release candidate 2 of Certified asterisk-18.9-cert8.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/certified-18.9-cert8-rc2
and
https://downloads.asterisk.org/pub/telephony/certified-asterisk

Repository: https://github.com/asterisk/asterisk
Tag: certified-18.9-cert8-rc2

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

Thank You!

Change Log for Release asterisk-certified-18.9-cert8-rc2

Summary:

Rename dialplan_functions.xml to dialplan_functions_doc.xml
openssl: Supress deprecation warnings from OpenSSL 3.0

User Notes:

Upgrade Notes:

Closed Issues:

None

asterisk - Asterisk Release certified-18.9-cert8-rc1

Published by asterisk-org-access-app[bot] 8 months ago

The Asterisk Development Team would like to announce
release candidate 1 of Certified asterisk-18.9-cert8.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/certified-18.9-cert8-rc1
and
https://downloads.asterisk.org/pub/telephony/certified-asterisk

Repository: https://github.com/asterisk/asterisk
Tag: certified-18.9-cert8-rc1

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

Thank You!

Change Log for Release asterisk-certified-18.9-cert8-rc1

Summary:

.github: Add force_cherry_pick option to Releaser
.github: Remove start_version from Releaser
app_chanspy: Add 'D' option for dual-channel audio
.github: Update github-script to v7 and fix a rest bug
manager.c: Fix regression due to using wrong free function.
doc: Remove obsolete CHANGES-staging directrory
MergeApproved.yml: Remove unneeded concurrency
SECURITY.md: Update with correct documentation URL
chan_pjsip: Add PJSIPHangup dialplan app and manager action
Remove files that are no longer updated
res_speech: allow speech to translate input channel
.github: PRSubmitActions: Fix adding reviewers to PR
.github: New PR Submit workflows
.github: New PR Submit workflows
res_stasis: signal when new command is queued
logger.h: Add ability to change the prefix on SCOPE_TRACE output
.github: Fix job prereqs in PROpenedUpdated
.github: Block PR tests until approved
Add libjwt to third-party
res_pjsip: update qualify_timeout documentation with DNS note
lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
cel: add publish user event helper
file.c: Add ability to search custom dir for sounds
make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
func_periodic_hook: Add hangup step to avoid timeout
func_periodic_hook: Don't truncate channel name
safe_asterisk: Change directory permissions to 755
variables: Add additional variable dialplan functions.
ari-stubs: Fix more local anchor references
ari-stubs: Fix more local anchor references
ari-stubs: Fix broken documentation anchors
rest-api: Updates for new documentation site
.github: Update workflow-application-token-action to v2
app_voicemail: Fix for loop declarations
download_externals: Fix a few version related issues
Remove .lastclean and .version from source control
.github: Use generic releaser
manager: Tolerate stasis messages with no channel snapshot.
audiohook: Unlock channel in mute if no audiohooks present.
app_queue: Add support for applying caller priority change immediately.
.github: Fix cherry-pick reminder issues
app.h: Move declaration of ast_getdata_result before its first use
doc: Remove obsolete CHANGES-staging and UPGRADE-staging
res_geolocation: Ensure required 'location_info' is present.
Adds manager actions to allow move/remove/forward individual messages in a par..
app_voicemail: add CLI commands for message manipulation
.github: Minor tweak to Asterisk Releaser
.github: Suppress cherry-pick reminder for some situations
Cleanup deleted files

User Notes:

app_chanspy: Add 'D' option for dual-channel audio

The ChanSpy application now accepts the 'D' option which
will interleave the spied audio within the outgoing frames. The
purpose of this is to allow the audio to be read as a Dual channel
stream with separate incoming and outgoing audio. Setting both the
'o' option and the 'D' option and results in the 'D' option being
ignored.
chan_pjsip: Add PJSIPHangup dialplan app and manager action

A new dialplan app PJSIPHangup and AMI action allows you
to hang up an unanswered incoming PJSIP call with a specific SIP
response code in the 400 -> 699 range.
res_speech: allow speech to translate input channel

res_speech now supports translation of an input channel
to a format supported by the speech provider, provided a translation
path is available between the source format and provider capabilites.
res_stasis: signal when new command is queued

Call setup times should be significantly improved
when using ARI.
lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS

You no longer need to select DEBUG_THREADS to use
DETECT_DEADLOCKS. This removes a significant amount of overhead
if you just want to detect possible deadlocks vs needing full
lock tracing.
file.c: Add ability to search custom dir for sounds

A new option "sounds_search_custom_dir" has been added to
asterisk.conf that allows asterisk to search
AST_DATA_DIR/sounds/custom for sounds files before searching the
standard AST_DATA_DIR/sounds/ directory.
make_buildopts_h, et. al. Allow adding all cflags to buildopts.h

The "Build Options" entry in the "core show settings"
CLI command has been renamed to "ABI related Build Options" and
a new entry named "All Build Options" has been added that shows
both breaking and non-breaking options.
variables: Add additional variable dialplan functions.

Four new dialplan functions have been added.
GLOBAL_DELETE and DELETE have been added which allows
the deletion of global and channel variables.
GLOBAL_EXISTS and VARIABLE_EXISTS have been added
which checks whether a global or channel variable has
been set.
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 par..

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>

Upgrade Notes:

Closed Issues:

#129: [bug]: res_speech_aeap: Crash due to NULL format on setup
#170: [improvement]: app_voicemail - add CLI commands to manipulate messages
#181: [improvement]: app_voicemail - add manager actions to display and manipulate messages
#200: [bug]: Regression: In app.h an enum is used before its declaration.
#202: [improvement]: app_queue: Add support for immediately applying queue caller priority change
#233: [bug]: Deadlock with MixMonitorMute AMI action
#263: [bug]: download_externals doesn't always handle versions correctly
#275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99 -Wdeclaration-after-statement'
#289: [new-feature]: Add support for deleting channel and global variables
#315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files before /var/lib/asterisk/sounds/
#316: [bug]: Privilege Escalation in Astrisk's Group permissions.
#319: [bug]: func_periodic_hook truncates long channel names when setting EncodedChannel
#321: [bug]: Performance suffers unnecessarily when debugging deadlocks
#325: [bug]: hangup after beep to avoid waiting for timeout
#330: [improvement]: Add cel user event helper function
#349: [improvement]: Add libjwt to third-party
#352: [bug]: Update qualify_timeout documentation to include DNS note
#360: [improvement]: Update documentation for CHANGES/UPGRADE files
#362: [improvement]: Speed up ARI command processing
#513: [bug]: manager.c: Crash due to regression using wrong free function when built with MALLOC_DEBUG
#569: [improvement]: Add option to interleave input and output frames on spied channel

asterisk - Asterisk Release 21.1.0

Published by asterisk-org-access-app[bot] 9 months ago

The Asterisk Development Team would like to announce
the release of asterisk-21.1.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/21.1.0
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-21.1.0

Summary:

logger: Fix linking regression.
Revert "core & res_pjsip: Improve topology change handling."
menuselect: Use more specific error message.
res_pjsip_nat: Fix potential use of uninitialized transport details
app_if: Fix faulty EndIf branching.
manager.c: Fix regression due to using wrong free function.
doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
config_options.c: Fix truncation of option descriptions.
manager.c: Improve clarity of "manager show connected".
make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
general: Fix broken links.
MergeApproved.yml: Remove unneeded concurrency
app_dial: Add option "j" to preserve initial stream topology of caller
pbx_config.c: Don't crash when unloading module.
ast_coredumper: Increase reliability
logger.c: Move LOG_GROUP documentation to dedicated XML file.
res_odbc.c: Allow concurrent access to request odbc connections
res_pjsip_header_funcs.c: Check URI parameter length before copying.
config.c: Log #exec include failures.
make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
app_voicemail.c: Completely resequence mailbox folders.
sig_analog: Fix channel leak when mwimonitor is enabled.
res_rtp_asterisk.c: Update for OpenSSL 3+.
alembic: Update list of TLS methods available on ps_transports.
func_channel: Expose previously unsettable options.
app.c: Allow ampersands in playback lists to be escaped.
uri.c: Simplify ast_uri_make_host_with_port()
func_curl.c: Remove CURLOPT() plaintext documentation.
res_http_websocket.c: Set hostname on client for certificate validation.
live_ast: Add astcachedir to generated asterisk.conf.
SECURITY.md: Update with correct documentation URL
func_lock: Add missing see-also refs to documentation.
app_followme.c: Grab reference on nativeformats before using it
configs: Improve documentation for bandwidth in iax.conf.
logger: Add channel-based filtering.
chan_iax2.c: Don't send unsanitized data to the logger.
codec_ilbc: Disable system ilbc if version >= 3.0.0
resource_channels.c: Explicit codec request when creating UnicastRTP.
doc: Update IP Quality of Service links.
chan_pjsip: Add PJSIPHangup dialplan app and manager action
chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
chan_dahdi: Warn if nonexistent cadence is requested.
stasis: Update the snapshot after setting the redirect
ari: Provide the caller ID RDNIS for the channels
main/utils: Implement ast_get_tid() for OpenBSD
res_rtp_asterisk.c: Fix runtime issue with LibreSSL
app_directory: Add ADSI support to Directory.
core_local: Fix local channel parsing with slashes.
Remove files that are no longer updated
app_voicemail: Add AMI event for mailbox PIN changes.
app_queue.c: Emit unpause reason with PauseQueueMember event.
bridge_simple: Suppress unchanged topology change requests
res_pjsip: Include cipher limit in config error message.
res_speech: allow speech to translate input channel
res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
api.wiki.mustache: Fix indentation in generated markdown
pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
configs: Fix typo in pjsip.conf.sample.
res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters
.github: PRSubmitActions: Fix adding reviewers to PR
.github: New PR Submit workflows
.github: New PR Submit workflows
res_stasis: signal when new command is queued
ari/stasis: Indicate progress before playback on a bridge
func_curl.c: Ensure channel is locked when manipulating datastores.
.github: Fix job prereqs in PROpenedUpdated
.github: Block PR tests until approved
.github: Use generic releaser
logger.h: Add ability to change the prefix on SCOPE_TRACE output
Add libjwt to third-party
res_pjsip: update qualify_timeout documentation with DNS note
chan_dahdi: Clarify scope of callgroup/pickupgroup.
func_json: Fix crashes for some types
res_speech_aeap: add aeap error handling
app_voicemail: Disable ADSI if unavailable.
codec_builtin: Use multiples of 20 for maximum_ms
lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
asterisk.c: Use the euid's home directory to read/write cli history
res_pjsip_transport_websocket: Prevent transport from being destroyed before message finishes.
cel: add publish user event helper
chan_console: Fix deadlock caused by unclean thread exit.
file.c: Add ability to search custom dir for sounds
chan_iax2: Improve authentication debugging.
res_rtp_asterisk: fix wrong counter management in ioqueue objects
res_pjsip_pubsub: Add body_type to test_handler for unit tests
make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
func_periodic_hook: Add hangup step to avoid timeout
res_stasis_recording.c: Save recording state when unmuted.
res_speech_aeap: check for null format on response
func_periodic_hook: Don't truncate channel name
safe_asterisk: Change directory permissions to 755
chan_rtp: Implement RTP glue for UnicastRTP channels
app_queue: periodic announcement configurable start time.
variables: Add additional variable dialplan functions.
Restore CHANGES and UPGRADE.txt to allow cherry-picks to work

User Notes:

app_dial: Add option "j" to preserve initial stream topology of caller

The option "j" is now available for the Dial application which
uses the initial stream topology of the caller to create the outgoing
channels.
logger: Add channel-based filtering.

The console log can now be filtered by
channels or groups of channels, using the
logger filter CLI commands.
chan_pjsip: Add PJSIPHangup dialplan app and manager action

A new dialplan app PJSIPHangup and AMI action allows you
to hang up an unanswered incoming PJSIP call with a specific SIP
response code in the 400 -> 699 range.
app_voicemail: Add AMI event for mailbox PIN changes.

The VoicemailPasswordChange event is
now emitted whenever a mailbox password is updated,
containing the mailbox information and the new
password.
Resolves: #398
res_speech: allow speech to translate input channel

res_speech now supports translation of an input channel
to a format supported by the speech provider, provided a translation
path is available between the source format and provider capabilites.
res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters

With this update, the PJSIP realm lengths have been extended
to support up to 255 characters.
res_stasis: signal when new command is queued

Call setup times should be significantly improved
when using ARI.
lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS

You no longer need to select DEBUG_THREADS to use
DETECT_DEADLOCKS. This removes a significant amount of overhead
if you just want to detect possible deadlocks vs needing full
lock tracing.
file.c: Add ability to search custom dir for sounds

A new option "sounds_search_custom_dir" has been added to
asterisk.conf that allows asterisk to search
AST_DATA_DIR/sounds/custom for sounds files before searching the
standard AST_DATA_DIR/sounds/ directory.
make_buildopts_h, et. al. Allow adding all cflags to buildopts.h

The "Build Options" entry in the "core show settings"
CLI command has been renamed to "ABI related Build Options" and
a new entry named "All Build Options" has been added that shows
both breaking and non-breaking options.
chan_rtp: Implement RTP glue for UnicastRTP channels

The dial string option 'g' was added to the UnicastRTP channel
which enables RTP glue and therefore native RTP bridges with those
channels.
app_queue: periodic announcement configurable start time.

Introduce a new queue configuration option called
'periodic-announce-startdelay' which will vary the normal (historic)
behavior of starting the periodic announcement cycle at
periodic-announce-frequency seconds after entering the queue to start
the periodic announcement cycle at period-announce-startdelay seconds
after joining the queue. The default behavior if this config option is
not set remains unchanged.
Signed-off-by: Jaco Kroon [email protected]
variables: Add additional variable dialplan functions.

Four new dialplan functions have been added.
GLOBAL_DELETE and DELETE have been added which allows
the deletion of global and channel variables.
GLOBAL_EXISTS and VARIABLE_EXISTS have been added
which checks whether a global or channel variable has
been set.

Upgrade Notes:

app.c: Allow ampersands in playback lists to be escaped.

Ampersands in URLs passed to the Playback(),
Background(), SpeechBackground(), Read(), Authenticate(), or
Queue() applications as filename arguments can now be escaped by
single quoting the filename. Additionally, this is also possible when
using the CONFBRIDGE dialplan function, or configuring various
features in confbridge.conf and queues.conf.
pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.

The dtls_rekey will be disabled if webrtc support is
requested on an endpoint. A warning will also be emitted.
res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters

As part of this update, the maximum allowable length
for PJSIP endpoints and relevant resources has been increased from
40 to 255 characters. To take advantage of this enhancement, it is
recommended to run the necessary procedures (e.g., Alembic) to
update your schemas.

Closed Issues:

#84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
#129: [bug]: res_speech_aeap: Crash due to NULL format on setup
#242: [new-feature]: logger: Allow filtering logs in CLI by channel
#248: [bug]: core_local: Local channels cannot have slashes in the destination
#260: [bug]: maxptime must be changed to multiples of 20
#286: [improvement]: chan_iax2: Improve authentication debugging
#289: [new-feature]: Add support for deleting channel and global variables
#294: [improvement]: chan_dahdi: Improve call pickup documentation
#298: [improvement]: chan_rtp: Implement RTP glue
#301: [bug]: Number of ICE TURN threads continually growing
#303: [bug]: SpeechBackground never exits
#308: [bug]: chan_console: Deadlock when hanging up console channels
#315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files before /var/lib/asterisk/sounds/
#316: [bug]: Privilege Escalation in Astrisk's Group permissions.
#319: [bug]: func_periodic_hook truncates long channel names when setting EncodedChannel
#321: [bug]: Performance suffers unnecessarily when debugging deadlocks
#325: [bug]: hangup after beep to avoid waiting for timeout
#330: [improvement]: Add cel user event helper function
#335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in build_resource_tree
#337: [bug]: asterisk.c: The CLI history file is written to the wrong directory in some cases
#341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
#345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved Functionality
#349: [improvement]: Add libjwt to third-party
#352: [bug]: Update qualify_timeout documentation to include DNS note
#354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
#356: [new-feature]: app_directory: Add ADSI support.
#360: [improvement]: Update documentation for CHANGES/UPGRADE files
#362: [improvement]: Speed up ARI command processing
#379: [bug]: Orphaned taskprocessors cause shutdown delays
#384: [bug]: Unnecessary re-INVITE after answer
#388: [bug]: Crash in app_followme.c due to not acquiring a reference to nativeformats
#396: [improvement]: res_pjsip: Specify max ciphers allowed if too many provided
#398: [new-feature]: app_voicemail: Add AMI event for password change
#409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
#423: [improvement]: func_lock: Add missing see-also refs
#425: [improvement]: configs: Improve documentation for bandwidth in iax.conf.sample
#428: [bug]: cli: Output is truncated from "config show help"
#430: [bug]: Fix broken links
#442: [bug]: func_channel: Some channel options are not settable
#445: [bug]: ast_coredumper isn't figuring out file locations properly in all cases
#458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
#462: [new-feature]: app_dial: Add new option to preserve initial stream topology of caller
#465: [improvement]: Change res_odbc connection pool request logic to not lock around blocking operations
#482: [improvement]: manager.c: Improve clarity of "manager show connected" output
#509: [bug]: res_pjsip: Crash when looking up transport state in use
#513: [bug]: manager.c: Crash due to regression using wrong free function when built with MALLOC_DEBUG
#520: [improvement]: menuselect: Use more specific error message.
#530: [bug]: bridge_channel.c: Stream topology change amplification with multiple layers of Local channels
#539: [bug]: Existence of logger.xml causes linking failure

asterisk - Asterisk Release 20.6.0

Published by asterisk-org-access-app[bot] 9 months ago

The Asterisk Development Team would like to announce
the release of asterisk-20.6.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/20.6.0
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-20.6.0

Summary:

logger: Fix linking regression.
Revert "core & res_pjsip: Improve topology change handling."
menuselect: Use more specific error message.
res_pjsip_nat: Fix potential use of uninitialized transport details
app_if: Fix faulty EndIf branching.
manager.c: Fix regression due to using wrong free function.
config_options.c: Fix truncation of option descriptions.
manager.c: Improve clarity of "manager show connected".
make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
general: Fix broken links.
MergeApproved.yml: Remove unneeded concurrency
app_dial: Add option "j" to preserve initial stream topology of caller
ast_coredumper: Increase reliability
logger.c: Move LOG_GROUP documentation to dedicated XML file.
res_odbc.c: Allow concurrent access to request odbc connections
res_pjsip_header_funcs.c: Check URI parameter length before copying.
config.c: Log #exec include failures.
make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
app_voicemail.c: Completely resequence mailbox folders.
sig_analog: Fix channel leak when mwimonitor is enabled.
res_rtp_asterisk.c: Update for OpenSSL 3+.
alembic: Update list of TLS methods available on ps_transports.
func_channel: Expose previously unsettable options.
app.c: Allow ampersands in playback lists to be escaped.
uri.c: Simplify ast_uri_make_host_with_port()
func_curl.c: Remove CURLOPT() plaintext documentation.
res_http_websocket.c: Set hostname on client for certificate validation.
live_ast: Add astcachedir to generated asterisk.conf.
SECURITY.md: Update with correct documentation URL
func_lock: Add missing see-also refs to documentation.
app_followme.c: Grab reference on nativeformats before using it
configs: Improve documentation for bandwidth in iax.conf.
logger: Add channel-based filtering.
chan_iax2.c: Don't send unsanitized data to the logger.
codec_ilbc: Disable system ilbc if version >= 3.0.0
resource_channels.c: Explicit codec request when creating UnicastRTP.
doc: Update IP Quality of Service links.
chan_pjsip: Add PJSIPHangup dialplan app and manager action
chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
chan_dahdi: Warn if nonexistent cadence is requested.
stasis: Update the snapshot after setting the redirect
ari: Provide the caller ID RDNIS for the channels
main/utils: Implement ast_get_tid() for OpenBSD
res_rtp_asterisk.c: Fix runtime issue with LibreSSL
app_directory: Add ADSI support to Directory.
core_local: Fix local channel parsing with slashes.
Remove files that are no longer updated
app_voicemail: Add AMI event for mailbox PIN changes.
app_queue.c: Emit unpause reason with PauseQueueMember event.
bridge_simple: Suppress unchanged topology change requests
res_pjsip: Include cipher limit in config error message.
res_speech: allow speech to translate input channel
res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
api.wiki.mustache: Fix indentation in generated markdown
pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
configs: Fix typo in pjsip.conf.sample.
res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters
.github: PRSubmitActions: Fix adding reviewers to PR
.github: New PR Submit workflows
.github: New PR Submit workflows
res_stasis: signal when new command is queued
ari/stasis: Indicate progress before playback on a bridge
func_curl.c: Ensure channel is locked when manipulating datastores.
.github: Fix job prereqs in PROpenedUpdated
.github: Block PR tests until approved
Update config.yml
logger.h: Add ability to change the prefix on SCOPE_TRACE output
Add libjwt to third-party
res_pjsip: update qualify_timeout documentation with DNS note
chan_dahdi: Clarify scope of callgroup/pickupgroup.
func_json: Fix crashes for some types
res_speech_aeap: add aeap error handling
app_voicemail: Disable ADSI if unavailable.
codec_builtin: Use multiples of 20 for maximum_ms
lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
asterisk.c: Use the euid's home directory to read/write cli history
res_pjsip_transport_websocket: Prevent transport from being destroyed before message finishes.
cel: add publish user event helper
chan_console: Fix deadlock caused by unclean thread exit.
file.c: Add ability to search custom dir for sounds
chan_iax2: Improve authentication debugging.
res_rtp_asterisk: fix wrong counter management in ioqueue objects
make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
func_periodic_hook: Add hangup step to avoid timeout
res_stasis_recording.c: Save recording state when unmuted.
res_speech_aeap: check for null format on response
func_periodic_hook: Don't truncate channel name
safe_asterisk: Change directory permissions to 755
chan_rtp: Implement RTP glue for UnicastRTP channels
app_queue: periodic announcement configurable start time.
variables: Add additional variable dialplan functions.
Restore CHANGES and UPGRADE.txt to allow cherry-picks to work

User Notes:

app_dial: Add option "j" to preserve initial stream topology of caller

The option "j" is now available for the Dial application which
uses the initial stream topology of the caller to create the outgoing
channels.
logger: Add channel-based filtering.

The console log can now be filtered by
channels or groups of channels, using the
logger filter CLI commands.
chan_pjsip: Add PJSIPHangup dialplan app and manager action

A new dialplan app PJSIPHangup and AMI action allows you
to hang up an unanswered incoming PJSIP call with a specific SIP
response code in the 400 -> 699 range.
app_voicemail: Add AMI event for mailbox PIN changes.

The VoicemailPasswordChange event is
now emitted whenever a mailbox password is updated,
containing the mailbox information and the new
password.
Resolves: #398
res_speech: allow speech to translate input channel

res_speech now supports translation of an input channel
to a format supported by the speech provider, provided a translation
path is available between the source format and provider capabilites.
res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters

With this update, the PJSIP realm lengths have been extended
to support up to 255 characters.
res_stasis: signal when new command is queued

Call setup times should be significantly improved
when using ARI.
lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS

You no longer need to select DEBUG_THREADS to use
DETECT_DEADLOCKS. This removes a significant amount of overhead
if you just want to detect possible deadlocks vs needing full
lock tracing.
file.c: Add ability to search custom dir for sounds

A new option "sounds_search_custom_dir" has been added to
asterisk.conf that allows asterisk to search
AST_DATA_DIR/sounds/custom for sounds files before searching the
standard AST_DATA_DIR/sounds/ directory.
make_buildopts_h, et. al. Allow adding all cflags to buildopts.h

The "Build Options" entry in the "core show settings"
CLI command has been renamed to "ABI related Build Options" and
a new entry named "All Build Options" has been added that shows
both breaking and non-breaking options.
chan_rtp: Implement RTP glue for UnicastRTP channels

The dial string option 'g' was added to the UnicastRTP channel
which enables RTP glue and therefore native RTP bridges with those
channels.
app_queue: periodic announcement configurable start time.

Introduce a new queue configuration option called
'periodic-announce-startdelay' which will vary the normal (historic)
behavior of starting the periodic announcement cycle at
periodic-announce-frequency seconds after entering the queue to start
the periodic announcement cycle at period-announce-startdelay seconds
after joining the queue. The default behavior if this config option is
not set remains unchanged.
Signed-off-by: Jaco Kroon [email protected]
variables: Add additional variable dialplan functions.

Four new dialplan functions have been added.
GLOBAL_DELETE and DELETE have been added which allows
the deletion of global and channel variables.
GLOBAL_EXISTS and VARIABLE_EXISTS have been added
which checks whether a global or channel variable has
been set.

Upgrade Notes:

app.c: Allow ampersands in playback lists to be escaped.

Ampersands in URLs passed to the Playback(),
Background(), SpeechBackground(), Read(), Authenticate(), or
Queue() applications as filename arguments can now be escaped by
single quoting the filename. Additionally, this is also possible when
using the CONFBRIDGE dialplan function, or configuring various
features in confbridge.conf and queues.conf.
pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.

The dtls_rekey will be disabled if webrtc support is
requested on an endpoint. A warning will also be emitted.
res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters

As part of this update, the maximum allowable length
for PJSIP endpoints and relevant resources has been increased from
40 to 255 characters. To take advantage of this enhancement, it is
recommended to run the necessary procedures (e.g., Alembic) to
update your schemas.

Closed Issues:

#84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
#129: [bug]: res_speech_aeap: Crash due to NULL format on setup
#242: [new-feature]: logger: Allow filtering logs in CLI by channel
#248: [bug]: core_local: Local channels cannot have slashes in the destination
#260: [bug]: maxptime must be changed to multiples of 20
#286: [improvement]: chan_iax2: Improve authentication debugging
#289: [new-feature]: Add support for deleting channel and global variables
#294: [improvement]: chan_dahdi: Improve call pickup documentation
#298: [improvement]: chan_rtp: Implement RTP glue
#301: [bug]: Number of ICE TURN threads continually growing
#303: [bug]: SpeechBackground never exits
#308: [bug]: chan_console: Deadlock when hanging up console channels
#315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files before /var/lib/asterisk/sounds/
#316: [bug]: Privilege Escalation in Astrisk's Group permissions.
#319: [bug]: func_periodic_hook truncates long channel names when setting EncodedChannel
#321: [bug]: Performance suffers unnecessarily when debugging deadlocks
#325: [bug]: hangup after beep to avoid waiting for timeout
#330: [improvement]: Add cel user event helper function
#337: [bug]: asterisk.c: The CLI history file is written to the wrong directory in some cases
#341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
#345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved Functionality
#349: [improvement]: Add libjwt to third-party
#352: [bug]: Update qualify_timeout documentation to include DNS note
#354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
#356: [new-feature]: app_directory: Add ADSI support.
#360: [improvement]: Update documentation for CHANGES/UPGRADE files
#362: [improvement]: Speed up ARI command processing
#379: [bug]: Orphaned taskprocessors cause shutdown delays
#384: [bug]: Unnecessary re-INVITE after answer
#388: [bug]: Crash in app_followme.c due to not acquiring a reference to nativeformats
#396: [improvement]: res_pjsip: Specify max ciphers allowed if too many provided
#398: [new-feature]: app_voicemail: Add AMI event for password change
#409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
#423: [improvement]: func_lock: Add missing see-also refs
#425: [improvement]: configs: Improve documentation for bandwidth in iax.conf.sample
#428: [bug]: cli: Output is truncated from "config show help"
#430: [bug]: Fix broken links
#442: [bug]: func_channel: Some channel options are not settable
#445: [bug]: ast_coredumper isn't figuring out file locations properly in all cases
#458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
#462: [new-feature]: app_dial: Add new option to preserve initial stream topology of caller
#465: [improvement]: Change res_odbc connection pool request logic to not lock around blocking operations
#482: [improvement]: manager.c: Improve clarity of "manager show connected" output
#509: [bug]: res_pjsip: Crash when looking up transport state in use
#513: [bug]: manager.c: Crash due to regression using wrong free function when built with MALLOC_DEBUG
#520: [improvement]: menuselect: Use more specific error message.
#530: [bug]: bridge_channel.c: Stream topology change amplification with multiple layers of Local channels
#539: [bug]: Existence of logger.xml causes linking failure

asterisk - Asterisk Release 18.21.0

Published by asterisk-org-access-app[bot] 9 months ago

The Asterisk Development Team would like to announce
the release of asterisk-18.21.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/18.21.0
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-18.21.0

Summary:

logger: Fix linking regression.
Revert "core & res_pjsip: Improve topology change handling."
menuselect: Use more specific error message.
res_pjsip_nat: Fix potential use of uninitialized transport details
app_if: Fix faulty EndIf branching.
manager.c: Fix regression due to using wrong free function.
config_options.c: Fix truncation of option descriptions.
manager.c: Improve clarity of "manager show connected".
make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
general: Fix broken links.
MergeApproved.yml: Remove unneeded concurrency
app_dial: Add option "j" to preserve initial stream topology of caller
ast_coredumper: Increase reliability
logger.c: Move LOG_GROUP documentation to dedicated XML file.
res_odbc.c: Allow concurrent access to request odbc connections
res_pjsip_header_funcs.c: Check URI parameter length before copying.
config.c: Log #exec include failures.
make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
app_voicemail.c: Completely resequence mailbox folders.
sig_analog: Fix channel leak when mwimonitor is enabled.
res_rtp_asterisk.c: Update for OpenSSL 3+.
alembic: Update list of TLS methods available on ps_transports.
func_channel: Expose previously unsettable options.
app.c: Allow ampersands in playback lists to be escaped.
uri.c: Simplify ast_uri_make_host_with_port()
func_curl.c: Remove CURLOPT() plaintext documentation.
res_http_websocket.c: Set hostname on client for certificate validation.
live_ast: Add astcachedir to generated asterisk.conf.
SECURITY.md: Update with correct documentation URL
func_lock: Add missing see-also refs to documentation.
app_followme.c: Grab reference on nativeformats before using it
configs: Improve documentation for bandwidth in iax.conf.
logger: Add channel-based filtering.
chan_iax2.c: Don't send unsanitized data to the logger.
codec_ilbc: Disable system ilbc if version >= 3.0.0
resource_channels.c: Explicit codec request when creating UnicastRTP.
doc: Update IP Quality of Service links.
chan_pjsip: Add PJSIPHangup dialplan app and manager action
chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
chan_dahdi: Warn if nonexistent cadence is requested.
stasis: Update the snapshot after setting the redirect
ari: Provide the caller ID RDNIS for the channels
main/utils: Implement ast_get_tid() for OpenBSD
res_rtp_asterisk.c: Fix runtime issue with LibreSSL
app_directory: Add ADSI support to Directory.
core_local: Fix local channel parsing with slashes.
Remove files that are no longer updated
app_voicemail: Add AMI event for mailbox PIN changes.
app_queue.c: Emit unpause reason with PauseQueueMember event.
bridge_simple: Suppress unchanged topology change requests
res_pjsip: Include cipher limit in config error message.
res_speech: allow speech to translate input channel
res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
api.wiki.mustache: Fix indentation in generated markdown
pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
configs: Fix typo in pjsip.conf.sample.
res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters
.github: PRSubmitActions: Fix adding reviewers to PR
.github: New PR Submit workflows
.github: New PR Submit workflows
res_stasis: signal when new command is queued
ari/stasis: Indicate progress before playback on a bridge
func_curl.c: Ensure channel is locked when manipulating datastores.
.github: Fix job prereqs in PROpenedUpdated
.github: Block PR tests until approved
logger.h: Add ability to change the prefix on SCOPE_TRACE output
Add libjwt to third-party
res_pjsip: update qualify_timeout documentation with DNS note
chan_dahdi: Clarify scope of callgroup/pickupgroup.
func_json: Fix crashes for some types
res_speech_aeap: add aeap error handling
app_voicemail: Disable ADSI if unavailable.
codec_builtin: Use multiples of 20 for maximum_ms
lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
asterisk.c: Use the euid's home directory to read/write cli history
res_pjsip_transport_websocket: Prevent transport from being destroyed before message finishes.
cel: add publish user event helper
chan_console: Fix deadlock caused by unclean thread exit.
file.c: Add ability to search custom dir for sounds
chan_iax2: Improve authentication debugging.
res_rtp_asterisk: fix wrong counter management in ioqueue objects
make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
func_periodic_hook: Add hangup step to avoid timeout
res_stasis_recording.c: Save recording state when unmuted.
res_speech_aeap: check for null format on response
func_periodic_hook: Don't truncate channel name
safe_asterisk: Change directory permissions to 755
chan_rtp: Implement RTP glue for UnicastRTP channels
app_queue: periodic announcement configurable start time.
variables: Add additional variable dialplan functions.
Restore CHANGES and UPGRADE.txt to allow cherry-picks to work

User Notes:

app_dial: Add option "j" to preserve initial stream topology of caller

The option "j" is now available for the Dial application which
uses the initial stream topology of the caller to create the outgoing
channels.
logger: Add channel-based filtering.

The console log can now be filtered by
channels or groups of channels, using the
logger filter CLI commands.
chan_pjsip: Add PJSIPHangup dialplan app and manager action

A new dialplan app PJSIPHangup and AMI action allows you
to hang up an unanswered incoming PJSIP call with a specific SIP
response code in the 400 -> 699 range.
app_voicemail: Add AMI event for mailbox PIN changes.

The VoicemailPasswordChange event is
now emitted whenever a mailbox password is updated,
containing the mailbox information and the new
password.
Resolves: #398
res_speech: allow speech to translate input channel

res_speech now supports translation of an input channel
to a format supported by the speech provider, provided a translation
path is available between the source format and provider capabilites.
res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters

With this update, the PJSIP realm lengths have been extended
to support up to 255 characters.
res_stasis: signal when new command is queued

Call setup times should be significantly improved
when using ARI.
lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS

You no longer need to select DEBUG_THREADS to use
DETECT_DEADLOCKS. This removes a significant amount of overhead
if you just want to detect possible deadlocks vs needing full
lock tracing.
file.c: Add ability to search custom dir for sounds

A new option "sounds_search_custom_dir" has been added to
asterisk.conf that allows asterisk to search
AST_DATA_DIR/sounds/custom for sounds files before searching the
standard AST_DATA_DIR/sounds/ directory.
make_buildopts_h, et. al. Allow adding all cflags to buildopts.h

The "Build Options" entry in the "core show settings"
CLI command has been renamed to "ABI related Build Options" and
a new entry named "All Build Options" has been added that shows
both breaking and non-breaking options.
chan_rtp: Implement RTP glue for UnicastRTP channels

The dial string option 'g' was added to the UnicastRTP channel
which enables RTP glue and therefore native RTP bridges with those
channels.
app_queue: periodic announcement configurable start time.

Introduce a new queue configuration option called
'periodic-announce-startdelay' which will vary the normal (historic)
behavior of starting the periodic announcement cycle at
periodic-announce-frequency seconds after entering the queue to start
the periodic announcement cycle at period-announce-startdelay seconds
after joining the queue. The default behavior if this config option is
not set remains unchanged.
Signed-off-by: Jaco Kroon [email protected]
variables: Add additional variable dialplan functions.

Four new dialplan functions have been added.
GLOBAL_DELETE and DELETE have been added which allows
the deletion of global and channel variables.
GLOBAL_EXISTS and VARIABLE_EXISTS have been added
which checks whether a global or channel variable has
been set.

Upgrade Notes:

app.c: Allow ampersands in playback lists to be escaped.

Ampersands in URLs passed to the Playback(),
Background(), SpeechBackground(), Read(), Authenticate(), or
Queue() applications as filename arguments can now be escaped by
single quoting the filename. Additionally, this is also possible when
using the CONFBRIDGE dialplan function, or configuring various
features in confbridge.conf and queues.conf.
pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.

The dtls_rekey will be disabled if webrtc support is
requested on an endpoint. A warning will also be emitted.
res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters

As part of this update, the maximum allowable length
for PJSIP endpoints and relevant resources has been increased from
40 to 255 characters. To take advantage of this enhancement, it is
recommended to run the necessary procedures (e.g., Alembic) to
update your schemas.

Closed Issues:

#84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
#129: [bug]: res_speech_aeap: Crash due to NULL format on setup
#242: [new-feature]: logger: Allow filtering logs in CLI by channel
#248: [bug]: core_local: Local channels cannot have slashes in the destination
#260: [bug]: maxptime must be changed to multiples of 20
#286: [improvement]: chan_iax2: Improve authentication debugging
#289: [new-feature]: Add support for deleting channel and global variables
#294: [improvement]: chan_dahdi: Improve call pickup documentation
#298: [improvement]: chan_rtp: Implement RTP glue
#301: [bug]: Number of ICE TURN threads continually growing
#303: [bug]: SpeechBackground never exits
#308: [bug]: chan_console: Deadlock when hanging up console channels
#315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files before /var/lib/asterisk/sounds/
#316: [bug]: Privilege Escalation in Astrisk's Group permissions.
#319: [bug]: func_periodic_hook truncates long channel names when setting EncodedChannel
#321: [bug]: Performance suffers unnecessarily when debugging deadlocks
#325: [bug]: hangup after beep to avoid waiting for timeout
#330: [improvement]: Add cel user event helper function
#337: [bug]: asterisk.c: The CLI history file is written to the wrong directory in some cases
#341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
#345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved Functionality
#349: [improvement]: Add libjwt to third-party
#352: [bug]: Update qualify_timeout documentation to include DNS note
#354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
#356: [new-feature]: app_directory: Add ADSI support.
#360: [improvement]: Update documentation for CHANGES/UPGRADE files
#362: [improvement]: Speed up ARI command processing
#379: [bug]: Orphaned taskprocessors cause shutdown delays
#384: [bug]: Unnecessary re-INVITE after answer
#388: [bug]: Crash in app_followme.c due to not acquiring a reference to nativeformats
#396: [improvement]: res_pjsip: Specify max ciphers allowed if too many provided
#398: [new-feature]: app_voicemail: Add AMI event for password change
#409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
#423: [improvement]: func_lock: Add missing see-also refs
#425: [improvement]: configs: Improve documentation for bandwidth in iax.conf.sample
#428: [bug]: cli: Output is truncated from "config show help"
#430: [bug]: Fix broken links
#442: [bug]: func_channel: Some channel options are not settable
#445: [bug]: ast_coredumper isn't figuring out file locations properly in all cases
#458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
#462: [new-feature]: app_dial: Add new option to preserve initial stream topology of caller
#465: [improvement]: Change res_odbc connection pool request logic to not lock around blocking operations
#482: [improvement]: manager.c: Improve clarity of "manager show connected" output
#509: [bug]: res_pjsip: Crash when looking up transport state in use
#513: [bug]: manager.c: Crash due to regression using wrong free function when built with MALLOC_DEBUG
#520: [improvement]: menuselect: Use more specific error message.
#530: [bug]: bridge_channel.c: Stream topology change amplification with multiple layers of Local channels
#539: [bug]: Existence of logger.xml causes linking failure

asterisk - Asterisk Release 21.1.0-rc2

Published by asterisk-org-access-app[bot] 9 months ago

The Asterisk Development Team would like to announce
release candidate 2 of asterisk-21.1.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/21.1.0-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 asterisk-21.1.0-rc2

Summary:

logger: Fix linking regression.

User Notes:

Upgrade Notes:

Closed Issues:

#539: [bug]: Existence of logger.xml causes linking failure

asterisk - Asterisk Release 20.6.0-rc2

Published by asterisk-org-access-app[bot] 9 months ago

The Asterisk Development Team would like to announce
release candidate 2 of asterisk-20.6.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/20.6.0-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 asterisk-20.6.0-rc2

Summary:

logger: Fix linking regression.

User Notes:

Upgrade Notes:

Closed Issues:

#539: [bug]: Existence of logger.xml causes linking failure

asterisk - Asterisk Release 18.21.0-rc2

Published by asterisk-org-access-app[bot] 9 months ago

The Asterisk Development Team would like to announce
release candidate 2 of asterisk-18.21.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/18.21.0-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 asterisk-18.21.0-rc2

Summary:

logger: Fix linking regression.

User Notes:

Upgrade Notes:

Closed Issues:

#539: [bug]: Existence of logger.xml causes linking failure

asterisk - Asterisk Release 21.1.0-rc1

Published by asterisk-org-access-app[bot] 9 months ago

The Asterisk Development Team would like to announce
release candidate 1 of asterisk-21.1.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/21.1.0-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-21.1.0-rc1

Summary:

Revert "core & res_pjsip: Improve topology change handling."
menuselect: Use more specific error message.
res_pjsip_nat: Fix potential use of uninitialized transport details
app_if: Fix faulty EndIf branching.
manager.c: Fix regression due to using wrong free function.
doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
config_options.c: Fix truncation of option descriptions.
manager.c: Improve clarity of "manager show connected".
make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
general: Fix broken links.
MergeApproved.yml: Remove unneeded concurrency
app_dial: Add option "j" to preserve initial stream topology of caller
pbx_config.c: Don't crash when unloading module.
ast_coredumper: Increase reliability
logger.c: Move LOG_GROUP documentation to dedicated XML file.
res_odbc.c: Allow concurrent access to request odbc connections
res_pjsip_header_funcs.c: Check URI parameter length before copying.
config.c: Log #exec include failures.
make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
app_voicemail.c: Completely resequence mailbox folders.
sig_analog: Fix channel leak when mwimonitor is enabled.
res_rtp_asterisk.c: Update for OpenSSL 3+.
alembic: Update list of TLS methods available on ps_transports.
func_channel: Expose previously unsettable options.
app.c: Allow ampersands in playback lists to be escaped.
uri.c: Simplify ast_uri_make_host_with_port()
func_curl.c: Remove CURLOPT() plaintext documentation.
res_http_websocket.c: Set hostname on client for certificate validation.
live_ast: Add astcachedir to generated asterisk.conf.
SECURITY.md: Update with correct documentation URL
func_lock: Add missing see-also refs to documentation.
app_followme.c: Grab reference on nativeformats before using it
configs: Improve documentation for bandwidth in iax.conf.
logger: Add channel-based filtering.
chan_iax2.c: Don't send unsanitized data to the logger.
codec_ilbc: Disable system ilbc if version >= 3.0.0
resource_channels.c: Explicit codec request when creating UnicastRTP.
doc: Update IP Quality of Service links.
chan_pjsip: Add PJSIPHangup dialplan app and manager action
chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
chan_dahdi: Warn if nonexistent cadence is requested.
stasis: Update the snapshot after setting the redirect
ari: Provide the caller ID RDNIS for the channels
main/utils: Implement ast_get_tid() for OpenBSD
res_rtp_asterisk.c: Fix runtime issue with LibreSSL
app_directory: Add ADSI support to Directory.
core_local: Fix local channel parsing with slashes.
Remove files that are no longer updated
app_voicemail: Add AMI event for mailbox PIN changes.
app_queue.c: Emit unpause reason with PauseQueueMember event.
bridge_simple: Suppress unchanged topology change requests
res_pjsip: Include cipher limit in config error message.
res_speech: allow speech to translate input channel
res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
api.wiki.mustache: Fix indentation in generated markdown
pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
configs: Fix typo in pjsip.conf.sample.
res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters
.github: PRSubmitActions: Fix adding reviewers to PR
.github: New PR Submit workflows
.github: New PR Submit workflows
res_stasis: signal when new command is queued
ari/stasis: Indicate progress before playback on a bridge
func_curl.c: Ensure channel is locked when manipulating datastores.
.github: Fix job prereqs in PROpenedUpdated
.github: Block PR tests until approved
.github: Use generic releaser
logger.h: Add ability to change the prefix on SCOPE_TRACE output
Add libjwt to third-party
res_pjsip: update qualify_timeout documentation with DNS note
chan_dahdi: Clarify scope of callgroup/pickupgroup.
func_json: Fix crashes for some types
res_speech_aeap: add aeap error handling
app_voicemail: Disable ADSI if unavailable.
codec_builtin: Use multiples of 20 for maximum_ms
lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
asterisk.c: Use the euid's home directory to read/write cli history
res_pjsip_transport_websocket: Prevent transport from being destroyed before message finishes.
cel: add publish user event helper
chan_console: Fix deadlock caused by unclean thread exit.
file.c: Add ability to search custom dir for sounds
chan_iax2: Improve authentication debugging.
res_rtp_asterisk: fix wrong counter management in ioqueue objects
res_pjsip_pubsub: Add body_type to test_handler for unit tests
make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
func_periodic_hook: Add hangup step to avoid timeout
res_stasis_recording.c: Save recording state when unmuted.
res_speech_aeap: check for null format on response
func_periodic_hook: Don't truncate channel name
safe_asterisk: Change directory permissions to 755
chan_rtp: Implement RTP glue for UnicastRTP channels
app_queue: periodic announcement configurable start time.
variables: Add additional variable dialplan functions.
Restore CHANGES and UPGRADE.txt to allow cherry-picks to work

User Notes:

app_dial: Add option "j" to preserve initial stream topology of caller

The option "j" is now available for the Dial application which
uses the initial stream topology of the caller to create the outgoing
channels.
logger: Add channel-based filtering.

The console log can now be filtered by
channels or groups of channels, using the
logger filter CLI commands.
chan_pjsip: Add PJSIPHangup dialplan app and manager action

A new dialplan app PJSIPHangup and AMI action allows you
to hang up an unanswered incoming PJSIP call with a specific SIP
response code in the 400 -> 699 range.
app_voicemail: Add AMI event for mailbox PIN changes.

The VoicemailPasswordChange event is
now emitted whenever a mailbox password is updated,
containing the mailbox information and the new
password.
Resolves: #398
res_speech: allow speech to translate input channel

res_speech now supports translation of an input channel
to a format supported by the speech provider, provided a translation
path is available between the source format and provider capabilites.
res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters

With this update, the PJSIP realm lengths have been extended
to support up to 255 characters.
res_stasis: signal when new command is queued

Call setup times should be significantly improved
when using ARI.
lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS

You no longer need to select DEBUG_THREADS to use
DETECT_DEADLOCKS. This removes a significant amount of overhead
if you just want to detect possible deadlocks vs needing full
lock tracing.
file.c: Add ability to search custom dir for sounds

A new option "sounds_search_custom_dir" has been added to
asterisk.conf that allows asterisk to search
AST_DATA_DIR/sounds/custom for sounds files before searching the
standard AST_DATA_DIR/sounds/ directory.
make_buildopts_h, et. al. Allow adding all cflags to buildopts.h

The "Build Options" entry in the "core show settings"
CLI command has been renamed to "ABI related Build Options" and
a new entry named "All Build Options" has been added that shows
both breaking and non-breaking options.
chan_rtp: Implement RTP glue for UnicastRTP channels

The dial string option 'g' was added to the UnicastRTP channel
which enables RTP glue and therefore native RTP bridges with those
channels.
app_queue: periodic announcement configurable start time.

Introduce a new queue configuration option called
'periodic-announce-startdelay' which will vary the normal (historic)
behavior of starting the periodic announcement cycle at
periodic-announce-frequency seconds after entering the queue to start
the periodic announcement cycle at period-announce-startdelay seconds
after joining the queue. The default behavior if this config option is
not set remains unchanged.
Signed-off-by: Jaco Kroon [email protected]
variables: Add additional variable dialplan functions.

Four new dialplan functions have been added.
GLOBAL_DELETE and DELETE have been added which allows
the deletion of global and channel variables.
GLOBAL_EXISTS and VARIABLE_EXISTS have been added
which checks whether a global or channel variable has
been set.

Upgrade Notes:

app.c: Allow ampersands in playback lists to be escaped.

Ampersands in URLs passed to the Playback(),
Background(), SpeechBackground(), Read(), Authenticate(), or
Queue() applications as filename arguments can now be escaped by
single quoting the filename. Additionally, this is also possible when
using the CONFBRIDGE dialplan function, or configuring various
features in confbridge.conf and queues.conf.
pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.

The dtls_rekey will be disabled if webrtc support is
requested on an endpoint. A warning will also be emitted.
res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters

As part of this update, the maximum allowable length
for PJSIP endpoints and relevant resources has been increased from
40 to 255 characters. To take advantage of this enhancement, it is
recommended to run the necessary procedures (e.g., Alembic) to
update your schemas.

Closed Issues:

#84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
#129: [bug]: res_speech_aeap: Crash due to NULL format on setup
#242: [new-feature]: logger: Allow filtering logs in CLI by channel
#248: [bug]: core_local: Local channels cannot have slashes in the destination
#260: [bug]: maxptime must be changed to multiples of 20
#286: [improvement]: chan_iax2: Improve authentication debugging
#289: [new-feature]: Add support for deleting channel and global variables
#294: [improvement]: chan_dahdi: Improve call pickup documentation
#298: [improvement]: chan_rtp: Implement RTP glue
#301: [bug]: Number of ICE TURN threads continually growing
#303: [bug]: SpeechBackground never exits
#308: [bug]: chan_console: Deadlock when hanging up console channels
#315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files before /var/lib/asterisk/sounds/
#316: [bug]: Privilege Escalation in Astrisk's Group permissions.
#319: [bug]: func_periodic_hook truncates long channel names when setting EncodedChannel
#321: [bug]: Performance suffers unnecessarily when debugging deadlocks
#325: [bug]: hangup after beep to avoid waiting for timeout
#330: [improvement]: Add cel user event helper function
#335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in build_resource_tree
#337: [bug]: asterisk.c: The CLI history file is written to the wrong directory in some cases
#341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
#345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved Functionality
#349: [improvement]: Add libjwt to third-party
#352: [bug]: Update qualify_timeout documentation to include DNS note
#354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
#356: [new-feature]: app_directory: Add ADSI support.
#360: [improvement]: Update documentation for CHANGES/UPGRADE files
#362: [improvement]: Speed up ARI command processing
#379: [bug]: Orphaned taskprocessors cause shutdown delays
#384: [bug]: Unnecessary re-INVITE after answer
#388: [bug]: Crash in app_followme.c due to not acquiring a reference to nativeformats
#396: [improvement]: res_pjsip: Specify max ciphers allowed if too many provided
#398: [new-feature]: app_voicemail: Add AMI event for password change
#409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
#423: [improvement]: func_lock: Add missing see-also refs
#425: [improvement]: configs: Improve documentation for bandwidth in iax.conf.sample
#428: [bug]: cli: Output is truncated from "config show help"
#430: [bug]: Fix broken links
#442: [bug]: func_channel: Some channel options are not settable
#445: [bug]: ast_coredumper isn't figuring out file locations properly in all cases
#458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
#462: [new-feature]: app_dial: Add new option to preserve initial stream topology of caller
#465: [improvement]: Change res_odbc connection pool request logic to not lock around blocking operations
#482: [improvement]: manager.c: Improve clarity of "manager show connected" output
#509: [bug]: res_pjsip: Crash when looking up transport state in use
#513: [bug]: manager.c: Crash due to regression using wrong free function when built with MALLOC_DEBUG
#520: [improvement]: menuselect: Use more specific error message.
#530: [bug]: bridge_channel.c: Stream topology change amplification with multiple layers of Local channels

asterisk - Asterisk Release 20.6.0-rc1

Published by asterisk-org-access-app[bot] 9 months ago

The Asterisk Development Team would like to announce
release candidate 1 of asterisk-20.6.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/20.6.0-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-20.6.0-rc1

Summary:

Revert "core & res_pjsip: Improve topology change handling."
menuselect: Use more specific error message.
res_pjsip_nat: Fix potential use of uninitialized transport details
app_if: Fix faulty EndIf branching.
manager.c: Fix regression due to using wrong free function.
config_options.c: Fix truncation of option descriptions.
manager.c: Improve clarity of "manager show connected".
make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
general: Fix broken links.
MergeApproved.yml: Remove unneeded concurrency
app_dial: Add option "j" to preserve initial stream topology of caller
ast_coredumper: Increase reliability
logger.c: Move LOG_GROUP documentation to dedicated XML file.
res_odbc.c: Allow concurrent access to request odbc connections
res_pjsip_header_funcs.c: Check URI parameter length before copying.
config.c: Log #exec include failures.
make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
app_voicemail.c: Completely resequence mailbox folders.
sig_analog: Fix channel leak when mwimonitor is enabled.
res_rtp_asterisk.c: Update for OpenSSL 3+.
alembic: Update list of TLS methods available on ps_transports.
func_channel: Expose previously unsettable options.
app.c: Allow ampersands in playback lists to be escaped.
uri.c: Simplify ast_uri_make_host_with_port()
func_curl.c: Remove CURLOPT() plaintext documentation.
res_http_websocket.c: Set hostname on client for certificate validation.
live_ast: Add astcachedir to generated asterisk.conf.
SECURITY.md: Update with correct documentation URL
func_lock: Add missing see-also refs to documentation.
app_followme.c: Grab reference on nativeformats before using it
configs: Improve documentation for bandwidth in iax.conf.
logger: Add channel-based filtering.
chan_iax2.c: Don't send unsanitized data to the logger.
codec_ilbc: Disable system ilbc if version >= 3.0.0
resource_channels.c: Explicit codec request when creating UnicastRTP.
doc: Update IP Quality of Service links.
chan_pjsip: Add PJSIPHangup dialplan app and manager action
chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
chan_dahdi: Warn if nonexistent cadence is requested.
stasis: Update the snapshot after setting the redirect
ari: Provide the caller ID RDNIS for the channels
main/utils: Implement ast_get_tid() for OpenBSD
res_rtp_asterisk.c: Fix runtime issue with LibreSSL
app_directory: Add ADSI support to Directory.
core_local: Fix local channel parsing with slashes.
Remove files that are no longer updated
app_voicemail: Add AMI event for mailbox PIN changes.
app_queue.c: Emit unpause reason with PauseQueueMember event.
bridge_simple: Suppress unchanged topology change requests
res_pjsip: Include cipher limit in config error message.
res_speech: allow speech to translate input channel
res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
api.wiki.mustache: Fix indentation in generated markdown
pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
configs: Fix typo in pjsip.conf.sample.
res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters
.github: PRSubmitActions: Fix adding reviewers to PR
.github: New PR Submit workflows
.github: New PR Submit workflows
res_stasis: signal when new command is queued
ari/stasis: Indicate progress before playback on a bridge
func_curl.c: Ensure channel is locked when manipulating datastores.
.github: Fix job prereqs in PROpenedUpdated
.github: Block PR tests until approved
Update config.yml
logger.h: Add ability to change the prefix on SCOPE_TRACE output
Add libjwt to third-party
res_pjsip: update qualify_timeout documentation with DNS note
chan_dahdi: Clarify scope of callgroup/pickupgroup.
func_json: Fix crashes for some types
res_speech_aeap: add aeap error handling
app_voicemail: Disable ADSI if unavailable.
codec_builtin: Use multiples of 20 for maximum_ms
lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
asterisk.c: Use the euid's home directory to read/write cli history
res_pjsip_transport_websocket: Prevent transport from being destroyed before message finishes.
cel: add publish user event helper
chan_console: Fix deadlock caused by unclean thread exit.
file.c: Add ability to search custom dir for sounds
chan_iax2: Improve authentication debugging.
res_rtp_asterisk: fix wrong counter management in ioqueue objects
make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
func_periodic_hook: Add hangup step to avoid timeout
res_stasis_recording.c: Save recording state when unmuted.
res_speech_aeap: check for null format on response
func_periodic_hook: Don't truncate channel name
safe_asterisk: Change directory permissions to 755
chan_rtp: Implement RTP glue for UnicastRTP channels
app_queue: periodic announcement configurable start time.
variables: Add additional variable dialplan functions.
Restore CHANGES and UPGRADE.txt to allow cherry-picks to work

User Notes:

app_dial: Add option "j" to preserve initial stream topology of caller

The option "j" is now available for the Dial application which
uses the initial stream topology of the caller to create the outgoing
channels.
logger: Add channel-based filtering.

The console log can now be filtered by
channels or groups of channels, using the
logger filter CLI commands.
chan_pjsip: Add PJSIPHangup dialplan app and manager action

A new dialplan app PJSIPHangup and AMI action allows you
to hang up an unanswered incoming PJSIP call with a specific SIP
response code in the 400 -> 699 range.
app_voicemail: Add AMI event for mailbox PIN changes.

The VoicemailPasswordChange event is
now emitted whenever a mailbox password is updated,
containing the mailbox information and the new
password.
Resolves: #398
res_speech: allow speech to translate input channel

res_speech now supports translation of an input channel
to a format supported by the speech provider, provided a translation
path is available between the source format and provider capabilites.
res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters

With this update, the PJSIP realm lengths have been extended
to support up to 255 characters.
res_stasis: signal when new command is queued

Call setup times should be significantly improved
when using ARI.
lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS

You no longer need to select DEBUG_THREADS to use
DETECT_DEADLOCKS. This removes a significant amount of overhead
if you just want to detect possible deadlocks vs needing full
lock tracing.
file.c: Add ability to search custom dir for sounds

A new option "sounds_search_custom_dir" has been added to
asterisk.conf that allows asterisk to search
AST_DATA_DIR/sounds/custom for sounds files before searching the
standard AST_DATA_DIR/sounds/ directory.
make_buildopts_h, et. al. Allow adding all cflags to buildopts.h

The "Build Options" entry in the "core show settings"
CLI command has been renamed to "ABI related Build Options" and
a new entry named "All Build Options" has been added that shows
both breaking and non-breaking options.
chan_rtp: Implement RTP glue for UnicastRTP channels

The dial string option 'g' was added to the UnicastRTP channel
which enables RTP glue and therefore native RTP bridges with those
channels.
app_queue: periodic announcement configurable start time.

Introduce a new queue configuration option called
'periodic-announce-startdelay' which will vary the normal (historic)
behavior of starting the periodic announcement cycle at
periodic-announce-frequency seconds after entering the queue to start
the periodic announcement cycle at period-announce-startdelay seconds
after joining the queue. The default behavior if this config option is
not set remains unchanged.
Signed-off-by: Jaco Kroon [email protected]
variables: Add additional variable dialplan functions.

Four new dialplan functions have been added.
GLOBAL_DELETE and DELETE have been added which allows
the deletion of global and channel variables.
GLOBAL_EXISTS and VARIABLE_EXISTS have been added
which checks whether a global or channel variable has
been set.

Upgrade Notes:

app.c: Allow ampersands in playback lists to be escaped.

Ampersands in URLs passed to the Playback(),
Background(), SpeechBackground(), Read(), Authenticate(), or
Queue() applications as filename arguments can now be escaped by
single quoting the filename. Additionally, this is also possible when
using the CONFBRIDGE dialplan function, or configuring various
features in confbridge.conf and queues.conf.
pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.

The dtls_rekey will be disabled if webrtc support is
requested on an endpoint. A warning will also be emitted.
res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters

As part of this update, the maximum allowable length
for PJSIP endpoints and relevant resources has been increased from
40 to 255 characters. To take advantage of this enhancement, it is
recommended to run the necessary procedures (e.g., Alembic) to
update your schemas.

Closed Issues:

#84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
#129: [bug]: res_speech_aeap: Crash due to NULL format on setup
#242: [new-feature]: logger: Allow filtering logs in CLI by channel
#248: [bug]: core_local: Local channels cannot have slashes in the destination
#260: [bug]: maxptime must be changed to multiples of 20
#286: [improvement]: chan_iax2: Improve authentication debugging
#289: [new-feature]: Add support for deleting channel and global variables
#294: [improvement]: chan_dahdi: Improve call pickup documentation
#298: [improvement]: chan_rtp: Implement RTP glue
#301: [bug]: Number of ICE TURN threads continually growing
#303: [bug]: SpeechBackground never exits
#308: [bug]: chan_console: Deadlock when hanging up console channels
#315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files before /var/lib/asterisk/sounds/
#316: [bug]: Privilege Escalation in Astrisk's Group permissions.
#319: [bug]: func_periodic_hook truncates long channel names when setting EncodedChannel
#321: [bug]: Performance suffers unnecessarily when debugging deadlocks
#325: [bug]: hangup after beep to avoid waiting for timeout
#330: [improvement]: Add cel user event helper function
#337: [bug]: asterisk.c: The CLI history file is written to the wrong directory in some cases
#341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
#345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved Functionality
#349: [improvement]: Add libjwt to third-party
#352: [bug]: Update qualify_timeout documentation to include DNS note
#354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
#356: [new-feature]: app_directory: Add ADSI support.
#360: [improvement]: Update documentation for CHANGES/UPGRADE files
#362: [improvement]: Speed up ARI command processing
#379: [bug]: Orphaned taskprocessors cause shutdown delays
#384: [bug]: Unnecessary re-INVITE after answer
#388: [bug]: Crash in app_followme.c due to not acquiring a reference to nativeformats
#396: [improvement]: res_pjsip: Specify max ciphers allowed if too many provided
#398: [new-feature]: app_voicemail: Add AMI event for password change
#409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
#423: [improvement]: func_lock: Add missing see-also refs
#425: [improvement]: configs: Improve documentation for bandwidth in iax.conf.sample
#428: [bug]: cli: Output is truncated from "config show help"
#430: [bug]: Fix broken links
#442: [bug]: func_channel: Some channel options are not settable
#445: [bug]: ast_coredumper isn't figuring out file locations properly in all cases
#458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
#462: [new-feature]: app_dial: Add new option to preserve initial stream topology of caller
#465: [improvement]: Change res_odbc connection pool request logic to not lock around blocking operations
#482: [improvement]: manager.c: Improve clarity of "manager show connected" output
#509: [bug]: res_pjsip: Crash when looking up transport state in use
#513: [bug]: manager.c: Crash due to regression using wrong free function when built with MALLOC_DEBUG
#520: [improvement]: menuselect: Use more specific error message.
#530: [bug]: bridge_channel.c: Stream topology change amplification with multiple layers of Local channels

asterisk - Asterisk Release 18.21.0-rc1

Published by asterisk-org-access-app[bot] 9 months ago

The Asterisk Development Team would like to announce
release candidate 1 of asterisk-18.21.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/18.21.0-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-18.21.0-rc1

Summary:

Revert "core & res_pjsip: Improve topology change handling."
menuselect: Use more specific error message.
res_pjsip_nat: Fix potential use of uninitialized transport details
app_if: Fix faulty EndIf branching.
manager.c: Fix regression due to using wrong free function.
config_options.c: Fix truncation of option descriptions.
manager.c: Improve clarity of "manager show connected".
make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
general: Fix broken links.
MergeApproved.yml: Remove unneeded concurrency
app_dial: Add option "j" to preserve initial stream topology of caller
ast_coredumper: Increase reliability
logger.c: Move LOG_GROUP documentation to dedicated XML file.
res_odbc.c: Allow concurrent access to request odbc connections
res_pjsip_header_funcs.c: Check URI parameter length before copying.
config.c: Log #exec include failures.
make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
app_voicemail.c: Completely resequence mailbox folders.
sig_analog: Fix channel leak when mwimonitor is enabled.
res_rtp_asterisk.c: Update for OpenSSL 3+.
alembic: Update list of TLS methods available on ps_transports.
func_channel: Expose previously unsettable options.
app.c: Allow ampersands in playback lists to be escaped.
uri.c: Simplify ast_uri_make_host_with_port()
func_curl.c: Remove CURLOPT() plaintext documentation.
res_http_websocket.c: Set hostname on client for certificate validation.
live_ast: Add astcachedir to generated asterisk.conf.
SECURITY.md: Update with correct documentation URL
func_lock: Add missing see-also refs to documentation.
app_followme.c: Grab reference on nativeformats before using it
configs: Improve documentation for bandwidth in iax.conf.
logger: Add channel-based filtering.
chan_iax2.c: Don't send unsanitized data to the logger.
codec_ilbc: Disable system ilbc if version >= 3.0.0
resource_channels.c: Explicit codec request when creating UnicastRTP.
doc: Update IP Quality of Service links.
chan_pjsip: Add PJSIPHangup dialplan app and manager action
chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
chan_dahdi: Warn if nonexistent cadence is requested.
stasis: Update the snapshot after setting the redirect
ari: Provide the caller ID RDNIS for the channels
main/utils: Implement ast_get_tid() for OpenBSD
res_rtp_asterisk.c: Fix runtime issue with LibreSSL
app_directory: Add ADSI support to Directory.
core_local: Fix local channel parsing with slashes.
Remove files that are no longer updated
app_voicemail: Add AMI event for mailbox PIN changes.
app_queue.c: Emit unpause reason with PauseQueueMember event.
bridge_simple: Suppress unchanged topology change requests
res_pjsip: Include cipher limit in config error message.
res_speech: allow speech to translate input channel
res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
api.wiki.mustache: Fix indentation in generated markdown
pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
configs: Fix typo in pjsip.conf.sample.
res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters
.github: PRSubmitActions: Fix adding reviewers to PR
.github: New PR Submit workflows
.github: New PR Submit workflows
res_stasis: signal when new command is queued
ari/stasis: Indicate progress before playback on a bridge
func_curl.c: Ensure channel is locked when manipulating datastores.
.github: Fix job prereqs in PROpenedUpdated
.github: Block PR tests until approved
logger.h: Add ability to change the prefix on SCOPE_TRACE output
Add libjwt to third-party
res_pjsip: update qualify_timeout documentation with DNS note
chan_dahdi: Clarify scope of callgroup/pickupgroup.
func_json: Fix crashes for some types
res_speech_aeap: add aeap error handling
app_voicemail: Disable ADSI if unavailable.
codec_builtin: Use multiples of 20 for maximum_ms
lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
asterisk.c: Use the euid's home directory to read/write cli history
res_pjsip_transport_websocket: Prevent transport from being destroyed before message finishes.
cel: add publish user event helper
chan_console: Fix deadlock caused by unclean thread exit.
file.c: Add ability to search custom dir for sounds
chan_iax2: Improve authentication debugging.
res_rtp_asterisk: fix wrong counter management in ioqueue objects
make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
func_periodic_hook: Add hangup step to avoid timeout
res_stasis_recording.c: Save recording state when unmuted.
res_speech_aeap: check for null format on response
func_periodic_hook: Don't truncate channel name
safe_asterisk: Change directory permissions to 755
chan_rtp: Implement RTP glue for UnicastRTP channels
app_queue: periodic announcement configurable start time.
variables: Add additional variable dialplan functions.
Restore CHANGES and UPGRADE.txt to allow cherry-picks to work

User Notes:

app_dial: Add option "j" to preserve initial stream topology of caller

The option "j" is now available for the Dial application which
uses the initial stream topology of the caller to create the outgoing
channels.
logger: Add channel-based filtering.

The console log can now be filtered by
channels or groups of channels, using the
logger filter CLI commands.
chan_pjsip: Add PJSIPHangup dialplan app and manager action

A new dialplan app PJSIPHangup and AMI action allows you
to hang up an unanswered incoming PJSIP call with a specific SIP
response code in the 400 -> 699 range.
app_voicemail: Add AMI event for mailbox PIN changes.

The VoicemailPasswordChange event is
now emitted whenever a mailbox password is updated,
containing the mailbox information and the new
password.
Resolves: #398
res_speech: allow speech to translate input channel

res_speech now supports translation of an input channel
to a format supported by the speech provider, provided a translation
path is available between the source format and provider capabilites.
res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters

With this update, the PJSIP realm lengths have been extended
to support up to 255 characters.
res_stasis: signal when new command is queued

Call setup times should be significantly improved
when using ARI.
lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS

You no longer need to select DEBUG_THREADS to use
DETECT_DEADLOCKS. This removes a significant amount of overhead
if you just want to detect possible deadlocks vs needing full
lock tracing.
file.c: Add ability to search custom dir for sounds

A new option "sounds_search_custom_dir" has been added to
asterisk.conf that allows asterisk to search
AST_DATA_DIR/sounds/custom for sounds files before searching the
standard AST_DATA_DIR/sounds/ directory.
make_buildopts_h, et. al. Allow adding all cflags to buildopts.h

The "Build Options" entry in the "core show settings"
CLI command has been renamed to "ABI related Build Options" and
a new entry named "All Build Options" has been added that shows
both breaking and non-breaking options.
chan_rtp: Implement RTP glue for UnicastRTP channels

The dial string option 'g' was added to the UnicastRTP channel
which enables RTP glue and therefore native RTP bridges with those
channels.
app_queue: periodic announcement configurable start time.

Introduce a new queue configuration option called
'periodic-announce-startdelay' which will vary the normal (historic)
behavior of starting the periodic announcement cycle at
periodic-announce-frequency seconds after entering the queue to start
the periodic announcement cycle at period-announce-startdelay seconds
after joining the queue. The default behavior if this config option is
not set remains unchanged.
Signed-off-by: Jaco Kroon [email protected]
variables: Add additional variable dialplan functions.

Four new dialplan functions have been added.
GLOBAL_DELETE and DELETE have been added which allows
the deletion of global and channel variables.
GLOBAL_EXISTS and VARIABLE_EXISTS have been added
which checks whether a global or channel variable has
been set.

Upgrade Notes:

app.c: Allow ampersands in playback lists to be escaped.

Ampersands in URLs passed to the Playback(),
Background(), SpeechBackground(), Read(), Authenticate(), or
Queue() applications as filename arguments can now be escaped by
single quoting the filename. Additionally, this is also possible when
using the CONFBRIDGE dialplan function, or configuring various
features in confbridge.conf and queues.conf.
pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.

The dtls_rekey will be disabled if webrtc support is
requested on an endpoint. A warning will also be emitted.
res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters

As part of this update, the maximum allowable length
for PJSIP endpoints and relevant resources has been increased from
40 to 255 characters. To take advantage of this enhancement, it is
recommended to run the necessary procedures (e.g., Alembic) to
update your schemas.

Closed Issues:

#84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
#129: [bug]: res_speech_aeap: Crash due to NULL format on setup
#242: [new-feature]: logger: Allow filtering logs in CLI by channel
#248: [bug]: core_local: Local channels cannot have slashes in the destination
#260: [bug]: maxptime must be changed to multiples of 20
#286: [improvement]: chan_iax2: Improve authentication debugging
#289: [new-feature]: Add support for deleting channel and global variables
#294: [improvement]: chan_dahdi: Improve call pickup documentation
#298: [improvement]: chan_rtp: Implement RTP glue
#301: [bug]: Number of ICE TURN threads continually growing
#303: [bug]: SpeechBackground never exits
#308: [bug]: chan_console: Deadlock when hanging up console channels
#315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files before /var/lib/asterisk/sounds/
#316: [bug]: Privilege Escalation in Astrisk's Group permissions.
#319: [bug]: func_periodic_hook truncates long channel names when setting EncodedChannel
#321: [bug]: Performance suffers unnecessarily when debugging deadlocks
#325: [bug]: hangup after beep to avoid waiting for timeout
#330: [improvement]: Add cel user event helper function
#337: [bug]: asterisk.c: The CLI history file is written to the wrong directory in some cases
#341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
#345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved Functionality
#349: [improvement]: Add libjwt to third-party
#352: [bug]: Update qualify_timeout documentation to include DNS note
#354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
#356: [new-feature]: app_directory: Add ADSI support.
#360: [improvement]: Update documentation for CHANGES/UPGRADE files
#362: [improvement]: Speed up ARI command processing
#379: [bug]: Orphaned taskprocessors cause shutdown delays
#384: [bug]: Unnecessary re-INVITE after answer
#388: [bug]: Crash in app_followme.c due to not acquiring a reference to nativeformats
#396: [improvement]: res_pjsip: Specify max ciphers allowed if too many provided
#398: [new-feature]: app_voicemail: Add AMI event for password change
#409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
#423: [improvement]: func_lock: Add missing see-also refs
#425: [improvement]: configs: Improve documentation for bandwidth in iax.conf.sample
#428: [bug]: cli: Output is truncated from "config show help"
#430: [bug]: Fix broken links
#442: [bug]: func_channel: Some channel options are not settable
#445: [bug]: ast_coredumper isn't figuring out file locations properly in all cases
#458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
#462: [new-feature]: app_dial: Add new option to preserve initial stream topology of caller
#465: [improvement]: Change res_odbc connection pool request logic to not lock around blocking operations
#482: [improvement]: manager.c: Improve clarity of "manager show connected" output
#509: [bug]: res_pjsip: Crash when looking up transport state in use
#513: [bug]: manager.c: Crash due to regression using wrong free function when built with MALLOC_DEBUG
#520: [improvement]: menuselect: Use more specific error message.
#530: [bug]: bridge_channel.c: Stream topology change amplification with multiple layers of Local channels

asterisk - Asterisk Release 21.0.2

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

The Asterisk Development Team would like to announce
the release of asterisk-21.0.2.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/21.0.2
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-21.0.2

Summary:

res_rtp_asterisk: Fix regression issues with DTLS client check

User Notes:

Upgrade Notes:

Closed Issues:

#500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
#503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when it shouldn't
#505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs created by ast_sockaddr_from_pj_sockaddr()

asterisk - Asterisk Release 20.5.2

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

The Asterisk Development Team would like to announce
the release of asterisk-20.5.2.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/20.5.2
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-20.5.2

Summary:

res_rtp_asterisk: Fix regression issues with DTLS client check

User Notes:

Upgrade Notes:

Closed Issues:

#500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
#503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when it shouldn't
#505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs created by ast_sockaddr_from_pj_sockaddr()

asterisk - Asterisk Release 18.20.2

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

The Asterisk Development Team would like to announce
the release of asterisk-18.20.2.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/18.20.2
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-18.20.2

Summary:

res_rtp_asterisk: Fix regression issues with DTLS client check

User Notes:

Upgrade Notes:

Closed Issues:

#500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
#503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when it shouldn't
#505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs created by ast_sockaddr_from_pj_sockaddr()

asterisk - Asterisk Release certified-18.9-cert7

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

The Asterisk Development Team would like to announce
the release of Certified asterisk-18.9-cert7.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/certified-18.9-cert7
and
https://downloads.asterisk.org/pub/telephony/certified-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-certified-18.9-cert7

Summary:

res_rtp_asterisk: Fix regression issues with DTLS client check

User Notes:

Upgrade Notes:

Closed Issues:

#500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
#503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when it shouldn't
#505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs created by ast_sockaddr_from_pj_sockaddr()

asterisk - Asterisk Release certified-18.9-cert6

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

The Asterisk Development Team would like to announce security release
Certified Asterisk 18.9-cert6.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/certified-18.9-cert6
and
https://downloads.asterisk.org/pub/telephony/certified-asterisk

The following security advisories were resolved in this release:

Change Log for Release asterisk-certified-18.9-cert6

Summary:

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

User Notes:

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'.
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.
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.
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.
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.
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.
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.

Upgrade Notes:

Closed Issues:

None

asterisk - Asterisk Release 21.0.1

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

The Asterisk Development Team would like to announce security release
Asterisk 21.0.1.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/21.0.1
and
https://downloads.asterisk.org/pub/telephony/asterisk

The following security advisories were resolved in this release:

Change Log for Release asterisk-21.0.1

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

Related Projects

pure-ftpd

Pure FTP server

09 Sep 2009 624

asterisk

Git SVN clone of Digium's Asterisk repository

11 Mar 2010 10

md5ha1

Tool for computing WWW-Digest MD5 HA1 string

31 Aug 2015 15

voip_patrol

VoIP signaling and media test automation

31 Aug 2016 113

asterisk

Change Log for Release asterisk-21.2.0-rc1

Links:

Summary:

User Notes:

app_dial: Add dial time for progress/ringing.

app_voicemail: Allow preventing mark messages as urgent.

Stir/Shaken Refactor

Upgrade bundled pjproject to 2.14.

res_pjsip_outbound_registration.c: Add User-Agent header override

app_speech_utils.c: Allow partial speech results.

app_chanspy: Add 'D' option for dual-channel audio

app_voicemail_odbc: remove macrocontext from voicemail_messages table

chan_dahdi: Allow MWI to be manually toggled on channels.

Upgrade Notes:

Stir/Shaken Refactor

app_voicemail_odbc: remove macrocontext from voicemail_messages table

Closed Issues:

Change Log for Release asterisk-20.7.0-rc1

Links:

Summary:

User Notes:

app_dial: Add dial time for progress/ringing.

app_voicemail: Allow preventing mark messages as urgent.

Stir/Shaken Refactor

Upgrade bundled pjproject to 2.14.

app_speech_utils.c: Allow partial speech results.

app_chanspy: Add 'D' option for dual-channel audio

chan_dahdi: Allow MWI to be manually toggled on channels.

Upgrade Notes:

Stir/Shaken Refactor

Closed Issues:

Change Log for Release asterisk-18.22.0-rc1

Links:

Summary:

User Notes:

app_dial: Add dial time for progress/ringing.

app_voicemail: Allow preventing mark messages as urgent.

Stir/Shaken Refactor

Upgrade bundled pjproject to 2.14.

app_speech_utils.c: Allow partial speech results.

app_chanspy: Add 'D' option for dual-channel audio

chan_dahdi: Allow MWI to be manually toggled on channels.

Upgrade Notes:

Stir/Shaken Refactor

Closed Issues:

Change Log for Release asterisk-certified-18.9-cert8-rc2

Links:

Summary:

User Notes:

Upgrade Notes:

Closed Issues:

Change Log for Release asterisk-certified-18.9-cert8-rc1

Links:

Summary:

User Notes:

app_chanspy: Add 'D' option for dual-channel audio

chan_pjsip: Add PJSIPHangup dialplan app and manager action

res_speech: allow speech to translate input channel

res_stasis: signal when new command is queued

lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS

file.c: Add ability to search custom dir for sounds

make_buildopts_h, et. al. Allow adding all cflags to buildopts.h

variables: Add additional variable dialplan functions.

app_queue: Add support for applying caller priority change immediately.

Adds manager actions to allow move/remove/forward individual messages in a par..

app_voicemail: add CLI commands for message manipulation

Upgrade Notes:

Closed Issues:

Change Log for Release asterisk-21.1.0

Links:

Summary:

User Notes:

app_dial: Add option "j" to preserve initial stream topology of caller

logger: Add channel-based filtering.

chan_pjsip: Add PJSIPHangup dialplan app and manager action

app_voicemail: Add AMI event for mailbox PIN changes.

res_speech: allow speech to translate input channel

res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters

res_stasis: signal when new command is queued