--- /dev/null
+
+Change Log for Release asterisk-18.20.0-rc1
+========================================
+
+Links:
+----------------------------------------
+
+ - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0-rc1.md)
+ - [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0-rc1)
+ - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0-rc1.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:
+----------------------------------------
+
+- ### 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.
+
+
projects / thirdparty / asterisk.git / commitdiff
