diff --git a/.version b/.version index a9d087399d..ca76c51934 100644 --- a/.version +++ b/.version @@ -1 +1 @@ -18.19.0 +18.20.0-rc1 diff --git a/CHANGES.md b/CHANGES.md index 06371c0ebe..f21ddd0b50 120000 --- a/CHANGES.md +++ b/CHANGES.md @@ -1 +1 @@ -ChangeLogs/ChangeLog-18.19.0.md \ No newline at end of file +ChangeLogs/ChangeLog-18.20.0-rc1.md \ No newline at end of file diff --git a/ChangeLogs/ChangeLog-18.20.0-rc1.md b/ChangeLogs/ChangeLog-18.20.0-rc1.md new file mode 100644 index 0000000000..3cc8b90b6c --- /dev/null +++ b/ChangeLogs/ChangeLog-18.20.0-rc1.md @@ -0,0 +1,745 @@ + +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 + Show contents of mailbox @ + voicemail remove + Remove message from in mailbox @ + voicemail move + Move message in mailbox & from to + voicemail forward + Forward message in mailbox @ to + mailbox @ + +- ### 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 + Show contents of mailbox @ + + voicemail remove + Remove message from in mailbox @ + + voicemail move + Move message in mailbox & from to + + voicemail forward + Forward message in mailbox @ to + mailbox @ + + +- ### 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. + + diff --git a/contrib/realtime/postgresql/postgresql_config.sql b/contrib/realtime/postgresql/postgresql_config.sql index 30fa3714d8..47e1a76cfe 100644 --- a/contrib/realtime/postgresql/postgresql_config.sql +++ b/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;