Update for 20.5.0-rc1

.version

-20.4.0
+20.5.0-rc1

CHANGES.md

-ChangeLogs/ChangeLog-20.4.0.md
\ No newline at end of file
+ChangeLogs/ChangeLog-20.5.0-rc1.md
\ No newline at end of file

ChangeLogs/ChangeLog-20.5.0-rc1.md

+
+Change Log for Release asterisk-20.5.0-rc1
+========================================
+
+Links:
+----------------------------------------
+
+ - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0-rc1.md)  
+ - [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0-rc1)  
+ - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0-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.
+
+

contrib/realtime/postgresql/postgresql_config.sql

@@ -1488,7 +1488,7 @@ UPDATE alembic_version SET version_num='9f3692b1654b' WHERE alembic_version.vers
 
 CREATE TYPE pjsip_100rel_values_v2 AS ENUM ('no', 'required', 'peer_supported', 'yes');
 
-ALTER TABLE ps_endpoints ALTER COLUMN 100rel TYPE pjsip_100rel_values_v2 USING 100rel::text::pjsip_100rel_values_v2;
+ALTER TABLE ps_endpoints ALTER COLUMN "100rel" TYPE pjsip_100rel_values_v2 USING "100rel"::text::pjsip_100rel_values_v2;
 
 DROP TYPE pjsip_100rel_values;