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asterisk/ChangeLogs/ChangeLog-20.5.0.md

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Change Log for Release asterisk-20.5.0
========================================
Links:
----------------------------------------
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
Summary:
----------------------------------------
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a particular mailbox folder. The forward command can be used to copy a message within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry, required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp->themssrc_valid` into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
User Notes:
----------------------------------------
- ### sig_analog: Add Called Subscriber Held capability.
Called Subscriber Held is now supported for analog
FXS channels, using the calledsubscriberheld option. This allows
a station user to go on hook when receiving an incoming call
and resume from another phone on the same line by going on hook,
without disconnecting the call.
- ### res_pjsip_header_funcs: Make prefix argument optional.
The prefix argument to PJSIP_HEADERS is now
optional. If not specified, all header names will be
returned.
- ### core/ari/pjsip: Add refer mechanism
There is a new ARI endpoint `/endpoints/refer` for referring
an endpoint to some URI or endpoint.
- ### chan_dahdi: Allow autoreoriginating after hangup.
The autoreoriginate setting now allows for kewlstart FXS
channels to automatically reoriginate and provide dial tone to the
user again after all calls on the line have cleared. This saves users
from having to manually hang up and pick up the receiver again before
making another call.
- ### sig_analog: Allow three-way flash to time out to silence.
The threewaysilenthold option now allows the three-way
dial tone to time out to silence, rather than continuing forever.
- ### res_pjsip: Enable TLS v1.3 if present.
res_pjsip now allows TLS v1.3 to be enabled if supported by
the underlying PJSIP library. The bundled version of PJSIP supports
TLS v1.3.
- ### app_queue: Add support for applying caller priority change immediately.
The 'queue priority caller' CLI command and
'QueueChangePriorityCaller' AMI action now have an 'immediate'
argument which allows the caller priority change to be reflected
immediately, causing the position of a caller to move within the
queue depending on the priorities of the other callers.
- ### Adds manager actions to allow move/remove/forward individual messages in a particular mailbox folder. The forward command can be used to copy a message within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry, required to retrieve message ID's.
The following manager actions have been added
VoicemailBoxSummary - Generate message list for a given mailbox
VoicemailRemove - Remove a message from a mailbox folder
VoicemailMove - Move a message from one folder to another within a mailbox
VoicemailForward - Copy a message from one folder in one mailbox
to another folder in another or the same mailbox.
- ### app_voicemail: add CLI commands for message manipulation
The following CLI commands have been added to app_voicemail
voicemail show mailbox <mailbox> <context>
Show contents of mailbox <mailbox>@<context>
voicemail remove <mailbox> <context> <from_folder> <messageid>
Remove message <messageid> from <from_folder> in mailbox <mailbox>@<context>
voicemail move <mailbox> <context> <from_folder> <messageid> <to_folder>
Move message <messageid> in mailbox <mailbox>&<context> from <from_folder> to <to_folder>
voicemail forward <from_mailbox> <from_context> <from_folder> <messageid> <to_mailbox> <to_context> <to_folder>
Forward message <messageid> in mailbox <mailbox>@<context> <from_folder> to
mailbox <mailbox>@<context> <to_folder>
- ### sig_analog: Allow immediate fake ring to be suppressed.
The immediatering option can now be set to no to suppress
the fake audible ringback provided when immediate=yes on FXS channels.
Upgrade Notes:
----------------------------------------
Closed Issues:
----------------------------------------
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99 -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while receiving digits
Commits By Author:
----------------------------------------
- ### Asterisk Development Team (1):
- Update for 20.5.0-rc1
- ### Bastian Triller (1):
- res_pjsip_session: Send Session Interval too small response
- ### George Joseph (12):
- .github: Suppress cherry-pick reminder for some situations
- .github: Minor tweak to Asterisk Releaser
- .github: Fix cherry-pick reminder issues
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- rest-api: Run make ari-stubs
- .github: Use generic releaser
- download_externals: Fix a few version related issues
- alembic: Fix quoting of the 100rel column
- .github: Update workflow-application-token-action to v2
- ari-stubs: Fix broken documentation anchors
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ### Holger Hans Peter Freyther (1):
- res_prometheus: Do not generate broken metrics
- ### Jason D. McCormick (1):
- install_prereq: Fix dependency install on aarch64.
- ### Joshua C. Colp (3):
- app_queue: Add support for applying caller priority change immediately.
- audiohook: Unlock channel in mute if no audiohooks present.
- manager: Tolerate stasis messages with no channel snapshot.
- ### Matthew Fredrickson (2):
- Revert "app_stack: Print proper exit location for PBXless channels."
- app_macro: Fix locking around datastore access
- ### Maximilian Fridrich (2):
- core/ari/pjsip: Add refer mechanism
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- ### Mike Bradeen (3):
- app_voicemail: add CLI commands for message manipulation
- Adds manager actions to allow move/remove/forward individual messages in a particular mailbox folder. The forward command can be used to copy a message within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry, required to retrieve message ID's.
- app_voicemail: Fix for loop declarations
- ### MikeNaso (1):
- res_pjsip.c: Set contact_user on incoming call local Contact header
- ### Naveen Albert (7):
- sig_analog: Allow immediate fake ring to be suppressed.
- sig_analog: Allow three-way flash to time out to silence.
- chan_dahdi: Allow autoreoriginating after hangup.
- res_pjsip_header_funcs: Make prefix argument optional.
- sig_analog: Add Called Subscriber Held capability.
- pbx.c: Fix gcc 12 compiler warning.
- app_dial: Fix infinite loop when sending digits.
- ### Sean Bright (6):
- res_geolocation: Ensure required 'location_info' is present.
- chan_iax2.c: Avoid crash with IAX2 switch support.
- func_export: Use correct function argument as variable name.
- extensions.conf.sample: Remove reference to missing context.
- res_pjsip: Enable TLS v1.3 if present.
- extconfig: Allow explicit DB result set ordering to be disabled.
- ### phoneben (1):
- func_cut: Add example to documentation.
- ### zhengsh (2):
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp->themssrc_valid` into the scope of the rtp_instance lock.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
Detail:
----------------------------------------
- ### ari-stubs: Fix more local anchor references
Author: George Joseph
Date: 2023-09-05
Also allow CreateDocs job to be run manually with default branches.
- ### ari-stubs: Fix more local anchor references
Author: George Joseph
Date: 2023-09-05
Also allow CreateDocs job to be run manually with default branches.
- ### ari-stubs: Fix broken documentation anchors
Author: George Joseph
Date: 2023-09-05
All of the links that reference page anchors with capital letters in
the ids (#Something) have been changed to lower case to match the
anchors that are generated by mkdocs.
- ### res_pjsip_session: Send Session Interval too small response
Author: Bastian Triller
Date: 2023-08-28
Handle session interval lower than endpoint's configured minimum timer
when sending first answer. Timer setting is checked during this step and
needs to handled appropriately.
Before this change, no response was sent at all. After this change a
response with 422 Session Interval too small is sent to UAC.
- ### .github: Update workflow-application-token-action to v2
Author: George Joseph
Date: 2023-08-31
- ### app_dial: Fix infinite loop when sending digits.
Author: Naveen Albert
Date: 2023-08-28
If the called party hangs up while digits are being
sent, -1 is returned to indicate so, but app_dial
was not checking the return value, resulting in
the hangup being lost and looping forever until
the caller manually hangs up the channel. We now
abort if digit sending fails.
ASTERISK-29428 #close
Resolves: #281
- ### app_voicemail: Fix for loop declarations
Author: Mike Bradeen
Date: 2023-08-29
Resolve for loop initial declarations added in cli changes.
Resolves: #275
- ### alembic: Fix quoting of the 100rel column
Author: George Joseph
Date: 2023-08-28
Add quoting around the ps_endpoints 100rel column in the ALTER
statements. Although alembic doesn't complain when generating
sql statements, postgresql does (rightly so).
Resolves: #274
- ### pbx.c: Fix gcc 12 compiler warning.
Author: Naveen Albert
Date: 2023-08-27
Resolves: #277
- ### app_audiosocket: Fixed timeout with -1 to avoid busy loop.
Author: zhengsh
Date: 2023-08-24
Resolves: asterisk#234
- ### download_externals: Fix a few version related issues
Author: George Joseph
Date: 2023-08-18
* Fixed issue with the script not parsing the new tag format for
certified releases. The format changed from certified/18.9-cert5
to certified-18.9-cert5.
* Fixed issue where the asterisk version wasn't being considered
when looking for cached versions.
Resolves: #263
- ### main/refer.c: Fix double free in refer_data_destructor + potential leak
Author: Maximilian Fridrich
Date: 2023-08-21
Resolves: #267
- ### sig_analog: Add Called Subscriber Held capability.
Author: Naveen Albert
Date: 2023-08-09
This adds support for Called Subscriber Held for FXS
lines, which allows users to go on hook when receiving
a call and resume the call later from another phone on
the same line, without disconnecting the call. This is
a convenience mechanism that most real PSTN telephone
switches support.
ASTERISK-30372 #close
Resolves: #240
UserNote: Called Subscriber Held is now supported for analog
FXS channels, using the calledsubscriberheld option. This allows
a station user to go on hook when receiving an incoming call
and resume from another phone on the same line by going on hook,
without disconnecting the call.
- ### app_macro: Fix locking around datastore access
Author: Matthew Fredrickson
Date: 2023-08-21
app_macro sometimes would crash due to datastore list corruption on the
channel because of lack of locking around find and create process for
the macro datastore. This patch locks the channel lock prior to protect
against this problem.
Resolves: #265
- ### Revert "app_stack: Print proper exit location for PBXless channels."
Author: Matthew Fredrickson
Date: 2023-08-10
This reverts commit 617dad4cba1513dddce87b8e95a61415fb587cf1.
apps/app_stack.c: Revert buggy gosub patch
This seems to break the case when a predial macro calls a gosub.
When the gosub calls return, the Return function outputs:
app_stack.c:423 return_exec: Return without Gosub: stack is empty
This returns -1 to the calling macro, which returns to app_dial
and causes the call to hangup instead of proceeding with the macro
that invoked the gosub.
Resolves: #253
- ### .github: Use generic releaser
Author: George Joseph
Date: 2023-08-15
- ### install_prereq: Fix dependency install on aarch64.
Author: Jason D. McCormick
Date: 2023-04-28
Fixes dependency solutions in install_prereq for Debian aarch64
platforms. install_prereq was attempting to forcibly install 32-bit
armhf packages due to the aptitude search for dependencies.
Resolves: #37
- ### res_pjsip.c: Set contact_user on incoming call local Contact header
Author: MikeNaso
Date: 2023-08-08
If the contact_user is configured on the endpoint it will now be set on the local Contact header URI for incoming calls. The contact_user has already been set on the local Contact header URI for outgoing calls.
Resolves: #226
- ### extconfig: Allow explicit DB result set ordering to be disabled.
Author: Sean Bright
Date: 2023-07-12
Added a new boolean configuration flag -
`order_multi_row_results_by_initial_column` - to both res_pgsql.conf
and res_config_odbc.conf that allows the administrator to disable the
explicit `ORDER BY` that was previously being added to all generated
SQL statements that returned multiple rows.
Fixes: #179
- ### rest-api: Run make ari-stubs
Author: George Joseph
Date: 2023-08-09
An earlier cherry-pick that involved rest-api somehow didn't include
a comment change in res/ari/resource_endpoints.h. This commit
corrects that. No changes other than the comment.
- ### res_pjsip_header_funcs: Make prefix argument optional.
Author: Naveen Albert
Date: 2023-08-09
The documentation for PJSIP_HEADERS claims that
prefix is optional, but in the code it is actually not.
However, there is no inherent reason for this, as users
may want to retrieve all header names, not just those
beginning with a certain prefix.
This makes the prefix optional for this function,
simply fetching all header names if not specified.
As a result, the documentation is now correct.
Resolves: #230
UserNote: The prefix argument to PJSIP_HEADERS is now
optional. If not specified, all header names will be
returned.
- ### pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
Author: George Joseph
Date: 2023-08-11
The default is 32 with 8 being used by pjproject itself. Recent
commits have put us over the limit resulting in assertions in
pjproject. Since this value is used in invites, dialogs,
transports and subscriptions as well as the global pjproject
endpoint, we don't want to increase it too much.
Resolves: #255
- ### manager: Tolerate stasis messages with no channel snapshot.
Author: Joshua C. Colp
Date: 2023-08-09
In some cases I have yet to determine some stasis messages may
be created without a channel snapshot. This change adds some
tolerance to this scenario, preventing a crash from occurring.
- ### core/ari/pjsip: Add refer mechanism
Author: Maximilian Fridrich
Date: 2023-05-10
This change adds support for refers that are not session based. It
includes a refer implementation for the PJSIP technology which results
in out-of-dialog REFERs being sent to a PJSIP endpoint. These can be
triggered using the new ARI endpoint `/endpoints/refer`.
Resolves: #71
UserNote: There is a new ARI endpoint `/endpoints/refer` for referring
an endpoint to some URI or endpoint.
- ### chan_dahdi: Allow autoreoriginating after hangup.
Author: Naveen Albert
Date: 2023-08-04
Currently, if an FXS channel is still off hook when
all calls on the line have hung up, the user is provided
reorder tone until going back on hook again.
In addition to not reflecting what most commercial switches
actually do, it's very common for switches to automatically
reoriginate for the user so that dial tone is provided without
the user having to depress and release the hookswitch manually.
This can increase convenience for users.
This behavior is now supported for kewlstart FXS channels.
It's supported only for kewlstart (FXOKS) mainly because the
behavior doesn't make any sense for ground start channels,
and loop start signalling doesn't provide the necessary DAHDI
event that makes this easy to implement. Likely almost everyone
is using FXOKS over FXOLS anyways since FXOLS is pretty useless
these days.
ASTERISK-30357 #close
Resolves: #224
UserNote: The autoreoriginate setting now allows for kewlstart FXS
channels to automatically reoriginate and provide dial tone to the
user again after all calls on the line have cleared. This saves users
from having to manually hang up and pick up the receiver again before
making another call.
- ### audiohook: Unlock channel in mute if no audiohooks present.
Author: Joshua C. Colp
Date: 2023-08-09
In the case where mute was called on a channel that had no
audiohooks the code was not unlocking the channel, resulting
in a deadlock.
Resolves: #233
- ### sig_analog: Allow three-way flash to time out to silence.
Author: Naveen Albert
Date: 2023-07-10
sig_analog allows users to flash and use the three-way dial
tone as a primitive hold function, simply by never timing
it out.
Some systems allow this dial tone to time out to silence,
so the user is not annoyed by a persistent dial tone.
This option allows the dial tone to time out normally to
silence.
ASTERISK-30004 #close
Resolves: #205
UserNote: The threewaysilenthold option now allows the three-way
dial tone to time out to silence, rather than continuing forever.
- ### res_prometheus: Do not generate broken metrics
Author: Holger Hans Peter Freyther
Date: 2023-04-07
In 8d6fdf9c3adede201f0ef026dab201b3a37b26b6 invisible bridges were
skipped but that lead to producing metrics with no name and no help.
Keep track of the number of metrics configured and then only emit these.
Add a basic testcase that verifies that there is no '(NULL)' in the
output.
ASTERISK-30474
- ### res_pjsip: Enable TLS v1.3 if present.
Author: Sean Bright
Date: 2023-08-02
Fixes #221
UserNote: res_pjsip now allows TLS v1.3 to be enabled if supported by
the underlying PJSIP library. The bundled version of PJSIP supports
TLS v1.3.
- ### func_cut: Add example to documentation.
Author: phoneben
Date: 2023-07-19
This adds an example to the XML documentation clarifying usage
of the CUT function to address a common misusage.
- ### extensions.conf.sample: Remove reference to missing context.
Author: Sean Bright
Date: 2023-07-16
c3ff4648 removed the [iaxtel700] context but neglected to remove
references to it.
This commit addresses that and also removes iaxtel and freeworlddialup
references from other config files.
- ### func_export: Use correct function argument as variable name.
Author: Sean Bright
Date: 2023-07-12
Fixes #208
- ### app_queue: Add support for applying caller priority change immediately.
Author: Joshua C. Colp
Date: 2023-07-07
The app_queue module provides both an AMI action and a CLI command
to change the priority of a caller in a queue. Up to now this change
of priority has only been reflected to new callers into the queue.
This change adds an "immediate" option to both the AMI action and
CLI command which immediately applies the priority change respective
to the other callers already in the queue. This can allow, for example,
a caller to be placed at the head of the queue immediately if their
priority is sufficient.
Resolves: #202
UserNote: 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.
- ### .github: Fix cherry-pick reminder issues
Author: George Joseph
Date: 2023-07-17
- ### chan_iax2.c: Avoid crash with IAX2 switch support.
Author: Sean Bright
Date: 2023-07-07
A change made in 82cebaa0 did not properly handle the case when a
channel was not provided, triggering a crash. ast_check_hangup(...)
does not protect against NULL pointers.
Fixes #180
- ### res_geolocation: Ensure required 'location_info' is present.
Author: Sean Bright
Date: 2023-07-07
Fixes #189
- ### Adds manager actions to allow move/remove/forward individual messages in a particular mailbox folder. The forward command can be used to copy a message within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry, required to retrieve message ID's.
Author: Mike Bradeen
Date: 2023-06-29
Resolves: #181
UserNote: 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
Author: Mike Bradeen
Date: 2023-06-20
Adds CLI commands to allow move/remove/forward individual messages
from a particular mailbox folder. The forward command can be used
to copy a message within a mailbox or to another mailbox. Also adds
a show mailbox, required to retrieve message ID's.
Resolves: #170
UserNote: The following CLI commands have been added to app_voicemail
voicemail show mailbox <mailbox> <context>
Show contents of mailbox <mailbox>@<context>
voicemail remove <mailbox> <context> <from_folder> <messageid>
Remove message <messageid> from <from_folder> in mailbox <mailbox>@<context>
voicemail move <mailbox> <context> <from_folder> <messageid> <to_folder>
Move message <messageid> in mailbox <mailbox>&<context> from <from_folder> to <to_folder>
voicemail forward <from_mailbox> <from_context> <from_folder> <messageid> <to_mailbox> <to_context> <to_folder>
Forward message <messageid> in mailbox <mailbox>@<context> <from_folder> to
mailbox <mailbox>@<context> <to_folder>
- ### res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp->themssrc_valid` into the scope of the rtp_instance lock.
Author: zhengsh
Date: 2023-06-30
From the gdb information, it was found that when calling __ast_free, the size of the
allocated space pointed to by the pointer matches the size created when rtp->themssrc_valid
is equal to 0. However, in reality, when reading the value of rtp->themssrc_valid in gdb,
it is found to be 1.
Within ast_rtcp_write(), the call to ast_rtp_rtcp_report_alloc() uses rtp->themssrc_valid,
which is outside the protection of the rtp_instance lock. However,
ast_rtcp_generate_report(), which is called by ast_rtcp_generate_compound_prefix(), uses
rtp->themssrc_valid within the protection of the rtp_instance lock.
This can lead to the possibility that the value of rtp->themssrc_valid used in the call to
ast_rtp_rtcp_report_alloc() may be different from the value of rtp->themssrc_valid used
within ast_rtcp_generate_report().
Resolves: asterisk#63
- ### .github: Minor tweak to Asterisk Releaser
Author: George Joseph
Date: 2023-07-12
- ### .github: Suppress cherry-pick reminder for some situations
Author: George Joseph
Date: 2023-07-11
In PROpenedOrUpdated, the cherry-pick reminder will now be
suppressed if there are already valid 'cherry-pick-to' comments
in the PR or the PR contained a 'cherry-pick-to: none' comment.
- ### sig_analog: Allow immediate fake ring to be suppressed.
Author: Naveen Albert
Date: 2023-06-08
When immediate=yes on an FXS channel, sig_analog will
start fake audible ringback that continues until the
channel is answered. Even if it answers immediately,
the ringback is still audible for a brief moment.
This can be disruptive and unwanted behavior.
This adds an option to disable this behavior, though
the default behavior remains unchanged.
ASTERISK-30003 #close
Resolves: #118
UserNote: The immediatering option can now be set to no to suppress
the fake audible ringback provided when immediate=yes on FXS channels.
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