A Dial timeout on POST /channels/{channelId}/dial will now result in a + CANCEL and ChannelDestroyed with cause 19 / User alerting, no answer. Previously + no explicit cause was set, resulting in a cause of 16 / Normal Call Clearing.
+Service and timer files for systemd have been added to the + contrib/systemd/ directory. If you are experiencing memory issues, + install these files to have "malloc trim" periodically run on the + system.
+This patch adds a global configuration option, log-caller-id-name, to queues.conf + to control whether the Caller ID name is logged as parameter 4 when a call enters a queue. + When log-caller-id-name=yes, the Caller ID name is included in the queue log, + Any '|' characters in the caller ID name will be replaced with '_'. + (provided it’s allowed by the existing log_restricted_caller_id rules). + When log-caller-id-name=no (the default), the Caller ID name is omitted.
+In cli.conf, you can now define startup commands that run before + core initialization and before module initialization.
+The AudioSocket protocol now forwards DTMF frames with + payload type 0x03. The payload is a 1-byte ascii representing the DTMF + digit (0-9,*,#...).
+res_pjproject: Fix DTLS client check failing on some platforms
+chan_pjsip: set correct Endpoint Device State on multiple channels
+file.c: missing "custom" sound files should not generate warning logs
+pbx_ael: unregister AELSub application and CLI commands on module load failure
+contrib: Add systemd service and timer files for malloc trim.
+audiosocket: added support for DTMF frames
+Prequisites for ARI Outbound Websockets
+channel: Always provide cause code in ChannelHangupRequest.
+docs: Fix typos in apps/
+chan_pjsip: Add the same details as PJSIPShowContacts to the CLI via 'pjsip s..
+stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
+res_pjsip_caller_id: Also parse URI parameters for ANI2.
+asterisk/channel.h: fix documentation for 'ast_waitfor_nandfds()'
+action_redirect: remove after_bridge_goto_info
+app_confbridge: Prevent crash when publishing channel-less event.
+Update config.guess and config.sub
+audiohook.c: Add ability to adjust volume with float
+Author: Naveen Albert + Date: 2025-04-26
+If the isup-oli was sent as a URI parameter, rather than a header + parameter, it was not being parsed. Make sure we parse both if + needed so the ANI2 is set regardless of which type of parameter + the isup-oli is sent as.
+Resolves: #1220
+Author: Naveen Albert + Date: 2025-04-26
+app_meetme is deprecated but wasn't removed as planned in 21, + so remove the inaccurate removal version.
+Resolves: #1224
+Author: Luz Paz + Date: 2025-04-09
+Found via codespell
+Author: Mike Bradeen + Date: 2025-04-17
+Other Dial operations (dial, app_dial) use Q.850 cause 19 when a dial timeout occurs, + but the Dial command via ARI did not set an explicit reason. This resulted in a + CANCEL with Normal Call Clearing and corresponding ChannelDestroyed.
+This change sets the hangup cause to AST_CAUSE_NO_ANSWER to be consistent with the + other operations.
+Fixes: #963
+UserNote: A Dial timeout on POST /channels/{channelId}/dial will now result in a + CANCEL and ChannelDestroyed with cause 19 / User alerting, no answer. Previously + no explicit cause was set, resulting in a cause of 16 / Normal Call Clearing.
+Author: Naveen Albert + Date: 2025-04-18
+Resolves: #1206
+Author: Andreas Wehrmann + Date: 2025-04-18
+This fixes crashes/hangs I noticed with Asterisk 20.3.0 and 20.13.0 and quickly found out, + that the AEL module doesn't do proper cleanup when it fails to load. + This happens for example when there are syntax errors and AEL fails to compile in which case pbx_load_module() + returns an error but load_module() doesn't then unregister CLI cmds and the application.
+Author: Albrecht Oster + Date: 2025-04-10
+Certain platforms (mainly BSD derivatives) have an additional length
+ field in sockaddr_in6 and sockaddr_in.
+ ast_sockaddr_from_pj_sockaddr() does not take this field into account
+ when copying over values from the pj_sockaddr into the ast_sockaddr.
+ The resulting ast_sockaddr will have an uninitialized value for
+ sin6_len/sin_len while the other ast_sockaddr (not converted from
+ a pj_sockaddr) to check against in ast_sockaddr_pj_sockaddr_cmp()
+ has the correct length value set.
This has the effect that ast_sockaddr_cmp() will always indicate
+ an address mismatch, because it does a bitwise comparison, and all DTLS
+ packets are dropped even if addresses and ports match.
ast_sockaddr_from_pj_sockaddr() now checks whether the length fields
+ are available on the current platform and sets the values accordingly.
Resolves: #505
+Author: George Joseph + Date: 2025-04-16
+stasis: + * Added stasis_app_is_registered(). + * Added stasis_app_control_mark_failed(). + * Added stasis_app_control_is_failed(). + * Fixed res_stasis_device_state so unsubscribe all works properly. + * Modified stasis_app_unregister() to unsubscribe from all event sources. + * Modified stasis_app_exec to return -1 if stasis_app_control_is_failed() + returns true.
+http: + * Added ast_http_create_basic_auth_header().
+md5: + * Added define for MD5_DIGEST_LENGTH.
+tcptls: + * Added flag to ast_tcptls_session_args to suppress connection log messages + to give callers more control over logging.
+http_websocket: + * Add flag to ast_websocket_client_options to suppress connection log messages + to give callers more control over logging. + * Added username and password to ast_websocket_client_options to support + outbound basic authentication. + * Added ast_websocket_result_to_str().
+Author: Ben Ford + Date: 2025-04-16
+Adds two files to the contrib/systemd/ directory that can be installed + to periodically run "malloc trim" on Asterisk. These files do nothing + unless they are explicitly moved to the correct location on the system. + Users who are experiencing Asterisk memory issues can use this service + to potentially help combat the problem. These files can also be + configured to change the start time and interval. See systemd.timer(5) + and systemd.time(7) for more information.
+UserNote: Service and timer files for systemd have been added to the + contrib/systemd/ directory. If you are experiencing memory issues, + install these files to have "malloc trim" periodically run on the + system.
+Author: Peter Jannesen + Date: 2025-03-13
+Under certain circumstances the context/extens/prio are stored in the + after_bridge_goto_info. This info is used when the bridge is broken by + for hangup of the other party. In the situation that the bridge is + broken by an AMI Redirect this info is not used but also not removed. + With the result that when the channel is put back in a bridge and the + bridge is broken the execution continues at the wrong + context/extens/prio.
+Resolves: #1144
+Author: Joshua C. Colp + Date: 2025-04-16
+When queueing a channel to be hung up a cause code can be + specified in one of two ways:
+ast_queue_hangup_with_cause + This function takes in a cause code and queues it as part + of the hangup request, which ultimately results in it being + set on the channel.
+ast_channel_hangupcause_set + ast_queue_hangup + This combination sets the hangup cause on the channel before + queueing the hangup instead of as part of that process.
+In the #2 case the ChannelHangupRequest event would not contain + the cause code. For consistency if a cause code has been set + on the channel it will now be added to the event.
+Resolves: #1197
+Author: phoneben + Date: 2025-02-28
+Add log-caller-id-name option to log Caller ID Name in queue log
+This patch introduces a new global configuration option, log-caller-id-name, + to queues.conf to control whether the Caller ID name is logged when a call enters a queue.
+When log-caller-id-name=yes, the Caller ID name is logged + as parameter 4 in the queue log, provided it’s allowed by the + existing log_restricted_caller_id rules. If log-caller-id-name=no (the default), + the Caller ID name is omitted from the logs.
+Fixes: #1091
+UserNote: This patch adds a global configuration option, log-caller-id-name, to queues.conf + to control whether the Caller ID name is logged as parameter 4 when a call enters a queue. + When log-caller-id-name=yes, the Caller ID name is included in the queue log, + Any '|' characters in the caller ID name will be replaced with '_'. + (provided it’s allowed by the existing log_restricted_caller_id rules). + When log-caller-id-name=no (the default), the Caller ID name is omitted.
+Author: George Joseph + Date: 2025-04-10
+Commands in the "[startup_commands]" section of cli.conf have historically run + after all core and module initialization has been completed and just before + "Asterisk Ready" is printed on the console. This meant that if you + wanted to debug initialization of a specific module, your only option + was to turn on debug for everything by setting "debug" in asterisk.conf.
+This commit introduces options to allow you to run CLI commands earlier in + the asterisk startup process.
+A command with a value of "pre-init" will run just after logger initialization + but before most core, and all module, initialization.
+A command with a value of "pre-module" will run just after all core + initialization but before all module initialization.
+A command with a value of "fully-booted" (or "yes" for backwards + compatibility) will run as they always have been...after all + initialization and just before "Asterisk Ready" is printed on the console.
+This means you could do this...
+[startup_commands]
+ core set debug 3 res_pjsip.so = pre-module
+ core set debug 0 res_pjsip.so = fully-booted
This would turn debugging on for res_pjsip.so to catch any module + initialization debug messages then turn it off again after the module is + loaded.
+UserNote: In cli.conf, you can now define startup commands that run before + core initialization and before module initialization.
+Author: Sean Bright + Date: 2025-04-07
+Resolves: #1190
+Author: George Joseph + Date: 2025-04-02
+ari_ws_session_registry_dtor() wasn't checking that the container was valid + before running ao2_callback on it to shutdown registered sessions.
+Author: George Joseph + Date: 2025-03-12
+This commit adds the ability to make ARI REST requests over the same + websocket used to receive events.
+For full details on how to use the new capability, visit...
+https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
+Changes:
+UpgradeNote: This commit adds the ability to make ARI REST requests over the same + websocket used to receive events. + See https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
+Author: mkmer + Date: 2025-03-18
+Add the capability to audiohook for float type volume adjustments. This allows for adjustments to volume smaller than 6dB. With INT adjustments, the first step is 2 which converts to ~6dB (or 1/2 volume / double volume depending on adjustment sign). 3dB is a typical adjustment level which can now be accommodated with an adjustment value of 1.41.
+This is accomplished by the following: + Convert internal variables to type float. + Always use ast_frame_adjust_volume_float() for adjustments. + Cast int to float in original functions ast_audiohook_volume_set(), and ast_volume_adjust(). + Cast float to int in ast_audiohook_volume_get() + Add functions ast_audiohook_volume_get_float, ast_audiohook_volume_set_float, and ast_audiohook_volume_adjust_float.
+This update maintains 100% backward compatibility.
+Resolves: #1171
+Author: Florent CHAUVEAU + Date: 2025-02-28
+Updated the AudioSocket protocol to allow sending DTMF frames. + AST_FRAME_DTMF frames are now forwarded to the server, in addition to + AST_FRAME_AUDIO frames. A new payload type AST_AUDIOSOCKET_KIND_DTMF + with value 0x03 was added to the protocol. The payload is a 1-byte + ascii representing the DTMF digit (0-9,*,#...).
+UserNote: The AudioSocket protocol now forwards DTMF frames with + payload type 0x03. The payload is a 1-byte ascii representing the DTMF + digit (0-9,*,#...).
+Author: Norm Harrison + Date: 2023-04-03
+Co-authored-by: Florent CHAUVEAU florentch@pm.me
+Author: Norm Harrison + Date: 2023-04-03
+Co-authored-by: Florent CHAUVEAU florentch@pm.me
+Author: Mark Murawski + Date: 2025-03-23
+CLI 'pjsip show contact' does not show enough information. + One must telnet to AMI or write a script to ask Asterisk for example what the User-Agent is on a Contact + This feature adds the same details as PJSIPShowContacts to the CLI
+Resolves: #643
+Author: Zhai Liangliang + Date: 2025-03-26
+Author: Alexei Gradinari + Date: 2025-03-25
+When one channel is placed on hold, the device state is set to ONHOLD + without checking other channels states. + In case of AST_CONTROL_HOLD set the device state as AST_DEVICE_UNKNOWN + to calculate aggregate device state of all active channels.
+The current implementation incorrectly classifies channels in use. + The only channels that has the states: UP, RING and BUSY are considered as "in use". + A channel should be considered "in use" if its state is anything other than + DOWN or RESERVED.
+Currently, if the number of channels "in use" is greater than device_state_busy_at, + the system does not set the state to BUSY. Instead, it incorrectly assigns an aggregate + device state. + The endpoint device state should be BUSY if the number of channels "in use" is greater + than or equal to device_state_busy_at.
+Fixes: #1181
+Author: Allan Nathanson + Date: 2025-03-18
+With sounds_search_custom_dir = yes we first look to see if a sound file
+ is present in the "custom" sound directory before looking in the standard
+ sound directories. We should not be issuing a WARNING log message if a
+ sound cannot be found in the "custom" directory.
Resolves: https://github.com/asterisk/asterisk/issues/1170
+ diff --git a/ChangeLogs/ChangeLog-21.9.0-rc1.md b/ChangeLogs/ChangeLog-21.9.0-rc1.md new file mode 100644 index 0000000000..d6ae81bac8 --- /dev/null +++ b/ChangeLogs/ChangeLog-21.9.0-rc1.md @@ -0,0 +1,564 @@ + +## Change Log for Release asterisk-21.9.0-rc1 + +### Links: + + - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0-rc1.html) + - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0-rc1) + - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0-rc1.tar.gz) + - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) + +### Summary: + +- Commits: 24 +- Commit Authors: 18 +- Issues Resolved: 12 +- Security Advisories Resolved: 0 + +### User Notes: + +- #### stasis/control.c: Set Hangup Cause to No Answer on Dial timeout + A Dial timeout on POST /channels/{channelId}/dial will now result in a + CANCEL and ChannelDestroyed with cause 19 / User alerting, no answer. Previously + no explicit cause was set, resulting in a cause of 16 / Normal Call Clearing. + +- #### contrib: Add systemd service and timer files for malloc trim. + Service and timer files for systemd have been added to the + contrib/systemd/ directory. If you are experiencing memory issues, + install these files to have "malloc trim" periodically run on the + system. + +- #### Add log-caller-id-name option to log Caller ID Name in queue log + This patch adds a global configuration option, log-caller-id-name, to queues.conf + to control whether the Caller ID name is logged as parameter 4 when a call enters a queue. + When log-caller-id-name=yes, the Caller ID name is included in the queue log, + Any '|' characters in the caller ID name will be replaced with '_'. + (provided it’s allowed by the existing log_restricted_caller_id rules). + When log-caller-id-name=no (the default), the Caller ID name is omitted. + +- #### asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf. + In cli.conf, you can now define startup commands that run before + core initialization and before module initialization. + +- #### audiosocket: added support for DTMF frames + The AudioSocket protocol now forwards DTMF frames with + payload type 0x03. The payload is a 1-byte ascii representing the DTMF + digit (0-9,*,#...). + + +### Upgrade Notes: + +- #### ARI: REST over Websocket + This commit adds the ability to make ARI REST requests over the same + websocket used to receive events. + See https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/ + + +### Commit Authors: + +- Albrecht Oster: (1) +- Alexei Gradinari: (1) +- Allan Nathanson: (1) +- Andreas Wehrmann: (1) +- Ben Ford: (1) +- Florent CHAUVEAU: (1) +- George Joseph: (4) +- Joshua C. Colp: (1) +- Luz Paz: (1) +- Mark Murawski: (1) +- Mike Bradeen: (1) +- Mkmer: (1) +- Naveen Albert: (3) +- Norm Harrison: (2) +- Peter Jannesen: (1) +- Phoneben: (1) +- Sean Bright: (1) +- Zhai Liangliang: (1) + +## Issue and Commit Detail: + +### Closed Issues: + + - 505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs created by ast_sockaddr_from_pj_sockaddr() + - 643: [new-feature]: pjsip show contact -- show all details same as AMI PJSIPShowContacts + - 963: [bug]: missing hangup cause for ARI ChannelDestroyed when Dial times out + - 1091: [improvement]: app queue :add to queue log callerid name + - 1144: [bug]: action_redirect don't remove bridge_after_goto data + - 1171: [improvement]: Need the capability in audiohook.c for fractional (float) type volume adjustments. + - 1181: [bug]: Incorrect PJSIP Endpoint Device States on Multiple Channels + - 1190: [bug]: Crash when starting ConfBridge recording over CLI and AMI + - 1197: [bug]: ChannelHangupRequest does not show cause code in all cases + - 1206: [improvement]: chan_iax2: Minor improvements to documentation and warning messages. + - 1220: [bug]: res_pjsip_caller_id: OLI is not parsed if contained in a URI parameter + - 1224: [improvement]: app_meetme: Removal version is incorrect + +### Commits By Author: + +- #### Albrecht Oster (1): + - res_pjproject: Fix DTLS client check failing on some platforms + +- #### Alexei Gradinari (1): + - chan_pjsip: set correct Endpoint Device State on multiple channels + +- #### Allan Nathanson (1): + - file.c: missing "custom" sound files should not generate warning logs + +- #### Andreas Wehrmann (1): + - pbx_ael: unregister AELSub application and CLI commands on module load failure + +- #### Ben Ford (1): + - contrib: Add systemd service and timer files for malloc trim. + +- #### Florent CHAUVEAU (1): + - audiosocket: added support for DTMF frames + +- #### George Joseph (4): + - ARI: REST over Websocket + - ari_websockets: Fix frack if ARI config fails to load. + - asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf. + - Prequisites for ARI Outbound Websockets + +- #### Joshua C. Colp (1): + - channel: Always provide cause code in ChannelHangupRequest. + +- #### Luz Paz (1): + - docs: Fix typos in apps/ + +- #### Mark Murawski (1): + - chan_pjsip: Add the same details as PJSIPShowContacts to the CLI via 'pjsip s.. + +- #### Mike Bradeen (1): + - stasis/control.c: Set Hangup Cause to No Answer on Dial timeout + +- #### Naveen Albert (3): + - chan_iax2: Minor improvements to documentation and warning messages. + - app_meetme: Remove inaccurate removal version from xmldocs. + - res_pjsip_caller_id: Also parse URI parameters for ANI2. + +- #### Norm Harrison (2): + - audiosocket: fix timeout, fix dialplan app exit, server address in logs + - asterisk/channel.h: fix documentation for 'ast_waitfor_nandfds()' + +- #### Peter Jannesen (1): + - action_redirect: remove after_bridge_goto_info + +- #### Sean Bright (1): + - app_confbridge: Prevent crash when publishing channel-less event. + +- #### Zhai Liangliang (1): + - Update config.guess and config.sub + +- #### mkmer (1): + - audiohook.c: Add ability to adjust volume with float + +- #### phoneben (1): + - Add log-caller-id-name option to log Caller ID Name in queue log + + +### Commit List: + +- res_pjsip_caller_id: Also parse URI parameters for ANI2. +- app_meetme: Remove inaccurate removal version from xmldocs. +- docs: Fix typos in apps/ +- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout +- chan_iax2: Minor improvements to documentation and warning messages. +- pbx_ael: unregister AELSub application and CLI commands on module load failure +- res_pjproject: Fix DTLS client check failing on some platforms +- Prequisites for ARI Outbound Websockets +- contrib: Add systemd service and timer files for malloc trim. +- action_redirect: remove after_bridge_goto_info +- channel: Always provide cause code in ChannelHangupRequest. +- Add log-caller-id-name option to log Caller ID Name in queue log +- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf. +- app_confbridge: Prevent crash when publishing channel-less event. +- ari_websockets: Fix frack if ARI config fails to load. +- ARI: REST over Websocket +- audiohook.c: Add ability to adjust volume with float +- audiosocket: added support for DTMF frames +- asterisk/channel.h: fix documentation for 'ast_waitfor_nandfds()' +- audiosocket: fix timeout, fix dialplan app exit, server address in logs +- Update config.guess and config.sub +- chan_pjsip: set correct Endpoint Device State on multiple channels +- file.c: missing "custom" sound files should not generate warning logs + +### Commit Details: + +#### res_pjsip_caller_id: Also parse URI parameters for ANI2. + Author: Naveen Albert + Date: 2025-04-26 + + If the isup-oli was sent as a URI parameter, rather than a header + parameter, it was not being parsed. Make sure we parse both if + needed so the ANI2 is set regardless of which type of parameter + the isup-oli is sent as. + + Resolves: #1220 + +#### app_meetme: Remove inaccurate removal version from xmldocs. + Author: Naveen Albert + Date: 2025-04-26 + + app_meetme is deprecated but wasn't removed as planned in 21, + so remove the inaccurate removal version. + + Resolves: #1224 + +#### docs: Fix typos in apps/ + Author: Luz Paz + Date: 2025-04-09 + + Found via codespell + + +#### stasis/control.c: Set Hangup Cause to No Answer on Dial timeout + Author: Mike Bradeen + Date: 2025-04-17 + + Other Dial operations (dial, app_dial) use Q.850 cause 19 when a dial timeout occurs, + but the Dial command via ARI did not set an explicit reason. This resulted in a + CANCEL with Normal Call Clearing and corresponding ChannelDestroyed. + + This change sets the hangup cause to AST_CAUSE_NO_ANSWER to be consistent with the + other operations. + + Fixes: #963 + + UserNote: A Dial timeout on POST /channels/{channelId}/dial will now result in a + CANCEL and ChannelDestroyed with cause 19 / User alerting, no answer. Previously + no explicit cause was set, resulting in a cause of 16 / Normal Call Clearing. + + +#### chan_iax2: Minor improvements to documentation and warning messages. + Author: Naveen Albert + Date: 2025-04-18 + + * Update Dial() documentation for IAX2 to include syntax for RSA + public key names. + * Add additional details to a couple warnings to provide more context + when an undecodable frame is received. + + Resolves: #1206 + +#### pbx_ael: unregister AELSub application and CLI commands on module load failure + Author: Andreas Wehrmann + Date: 2025-04-18 + + This fixes crashes/hangs I noticed with Asterisk 20.3.0 and 20.13.0 and quickly found out, + that the AEL module doesn't do proper cleanup when it fails to load. + This happens for example when there are syntax errors and AEL fails to compile in which case pbx_load_module() + returns an error but load_module() doesn't then unregister CLI cmds and the application. + + +#### res_pjproject: Fix DTLS client check failing on some platforms + Author: Albrecht Oster + Date: 2025-04-10 + + Certain platforms (mainly BSD derivatives) have an additional length + field in `sockaddr_in6` and `sockaddr_in`. + `ast_sockaddr_from_pj_sockaddr()` does not take this field into account + when copying over values from the `pj_sockaddr` into the `ast_sockaddr`. + The resulting `ast_sockaddr` will have an uninitialized value for + `sin6_len`/`sin_len` while the other `ast_sockaddr` (not converted from + a `pj_sockaddr`) to check against in `ast_sockaddr_pj_sockaddr_cmp()` + has the correct length value set. + + This has the effect that `ast_sockaddr_cmp()` will always indicate + an address mismatch, because it does a bitwise comparison, and all DTLS + packets are dropped even if addresses and ports match. + + `ast_sockaddr_from_pj_sockaddr()` now checks whether the length fields + are available on the current platform and sets the values accordingly. + + Resolves: #505 + +#### Prequisites for ARI Outbound Websockets + Author: George Joseph + Date: 2025-04-16 + + stasis: + * Added stasis_app_is_registered(). + * Added stasis_app_control_mark_failed(). + * Added stasis_app_control_is_failed(). + * Fixed res_stasis_device_state so unsubscribe all works properly. + * Modified stasis_app_unregister() to unsubscribe from all event sources. + * Modified stasis_app_exec to return -1 if stasis_app_control_is_failed() + returns true. + + http: + * Added ast_http_create_basic_auth_header(). + + md5: + * Added define for MD5_DIGEST_LENGTH. + + tcptls: + * Added flag to ast_tcptls_session_args to suppress connection log messages + to give callers more control over logging. + + http_websocket: + * Add flag to ast_websocket_client_options to suppress connection log messages + to give callers more control over logging. + * Added username and password to ast_websocket_client_options to support + outbound basic authentication. + * Added ast_websocket_result_to_str(). + + +#### contrib: Add systemd service and timer files for malloc trim. + Author: Ben Ford + Date: 2025-04-16 + + Adds two files to the contrib/systemd/ directory that can be installed + to periodically run "malloc trim" on Asterisk. These files do nothing + unless they are explicitly moved to the correct location on the system. + Users who are experiencing Asterisk memory issues can use this service + to potentially help combat the problem. These files can also be + configured to change the start time and interval. See systemd.timer(5) + and systemd.time(7) for more information. + + UserNote: Service and timer files for systemd have been added to the + contrib/systemd/ directory. If you are experiencing memory issues, + install these files to have "malloc trim" periodically run on the + system. + + +#### action_redirect: remove after_bridge_goto_info + Author: Peter Jannesen + Date: 2025-03-13 + + Under certain circumstances the context/extens/prio are stored in the + after_bridge_goto_info. This info is used when the bridge is broken by + for hangup of the other party. In the situation that the bridge is + broken by an AMI Redirect this info is not used but also not removed. + With the result that when the channel is put back in a bridge and the + bridge is broken the execution continues at the wrong + context/extens/prio. + + Resolves: #1144 + +#### channel: Always provide cause code in ChannelHangupRequest. + Author: Joshua C. Colp + Date: 2025-04-16 + + When queueing a channel to be hung up a cause code can be + specified in one of two ways: + + 1. ast_queue_hangup_with_cause + This function takes in a cause code and queues it as part + of the hangup request, which ultimately results in it being + set on the channel. + + 2. ast_channel_hangupcause_set + ast_queue_hangup + This combination sets the hangup cause on the channel before + queueing the hangup instead of as part of that process. + + In the #2 case the ChannelHangupRequest event would not contain + the cause code. For consistency if a cause code has been set + on the channel it will now be added to the event. + + Resolves: #1197 + +#### Add log-caller-id-name option to log Caller ID Name in queue log + Author: phoneben + Date: 2025-02-28 + + Add log-caller-id-name option to log Caller ID Name in queue log + + This patch introduces a new global configuration option, log-caller-id-name, + to queues.conf to control whether the Caller ID name is logged when a call enters a queue. + + When log-caller-id-name=yes, the Caller ID name is logged + as parameter 4 in the queue log, provided it’s allowed by the + existing log_restricted_caller_id rules. If log-caller-id-name=no (the default), + the Caller ID name is omitted from the logs. + + Fixes: #1091 + + UserNote: This patch adds a global configuration option, log-caller-id-name, to queues.conf + to control whether the Caller ID name is logged as parameter 4 when a call enters a queue. + When log-caller-id-name=yes, the Caller ID name is included in the queue log, + Any '|' characters in the caller ID name will be replaced with '_'. + (provided it’s allowed by the existing log_restricted_caller_id rules). + When log-caller-id-name=no (the default), the Caller ID name is omitted. + + +#### asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf. + Author: George Joseph + Date: 2025-04-10 + + Commands in the "[startup_commands]" section of cli.conf have historically run + after all core and module initialization has been completed and just before + "Asterisk Ready" is printed on the console. This meant that if you + wanted to debug initialization of a specific module, your only option + was to turn on debug for everything by setting "debug" in asterisk.conf. + + This commit introduces options to allow you to run CLI commands earlier in + the asterisk startup process. + + A command with a value of "pre-init" will run just after logger initialization + but before most core, and all module, initialization. + + A command with a value of "pre-module" will run just after all core + initialization but before all module initialization. + + A command with a value of "fully-booted" (or "yes" for backwards + compatibility) will run as they always have been...after all + initialization and just before "Asterisk Ready" is printed on the console. + + This means you could do this... + + ``` + [startup_commands] + core set debug 3 res_pjsip.so = pre-module + core set debug 0 res_pjsip.so = fully-booted + ``` + + This would turn debugging on for res_pjsip.so to catch any module + initialization debug messages then turn it off again after the module is + loaded. + + UserNote: In cli.conf, you can now define startup commands that run before + core initialization and before module initialization. + + +#### app_confbridge: Prevent crash when publishing channel-less event. + Author: Sean Bright + Date: 2025-04-07 + + Resolves: #1190 + +#### ari_websockets: Fix frack if ARI config fails to load. + Author: George Joseph + Date: 2025-04-02 + + ari_ws_session_registry_dtor() wasn't checking that the container was valid + before running ao2_callback on it to shutdown registered sessions. + + +#### ARI: REST over Websocket + Author: George Joseph + Date: 2025-03-12 + + This commit adds the ability to make ARI REST requests over the same + websocket used to receive events. + + For full details on how to use the new capability, visit... + + https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/ + + Changes: + + * Added utilities to http.c: + * ast_get_http_method_from_string(). + * ast_http_parse_post_form(). + * Added utilities to json.c: + * ast_json_nvp_array_to_ast_variables(). + * ast_variables_to_json_nvp_array(). + * Added definitions for new events to carry REST responses. + * Created res/ari/ari_websocket_requests.c to house the new request handlers. + * Moved non-event specific code out of res/ari/resource_events.c into + res/ari/ari_websockets.c + * Refactored res/res_ari.c to move non-http code out of ast_ari_callback() + (which is http specific) and into ast_ari_invoke() so it can be shared + between both the http and websocket transports. + + UpgradeNote: This commit adds the ability to make ARI REST requests over the same + websocket used to receive events. + See https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/ + + +#### audiohook.c: Add ability to adjust volume with float + Author: mkmer + Date: 2025-03-18 + + Add the capability to audiohook for float type volume adjustments. This allows for adjustments to volume smaller than 6dB. With INT adjustments, the first step is 2 which converts to ~6dB (or 1/2 volume / double volume depending on adjustment sign). 3dB is a typical adjustment level which can now be accommodated with an adjustment value of 1.41. + + This is accomplished by the following: + Convert internal variables to type float. + Always use ast_frame_adjust_volume_float() for adjustments. + Cast int to float in original functions ast_audiohook_volume_set(), and ast_volume_adjust(). + Cast float to int in ast_audiohook_volume_get() + Add functions ast_audiohook_volume_get_float, ast_audiohook_volume_set_float, and ast_audiohook_volume_adjust_float. + + This update maintains 100% backward compatibility. + + Resolves: #1171 + +#### audiosocket: added support for DTMF frames + Author: Florent CHAUVEAU + Date: 2025-02-28 + + Updated the AudioSocket protocol to allow sending DTMF frames. + AST_FRAME_DTMF frames are now forwarded to the server, in addition to + AST_FRAME_AUDIO frames. A new payload type AST_AUDIOSOCKET_KIND_DTMF + with value 0x03 was added to the protocol. The payload is a 1-byte + ascii representing the DTMF digit (0-9,*,#...). + + UserNote: The AudioSocket protocol now forwards DTMF frames with + payload type 0x03. The payload is a 1-byte ascii representing the DTMF + digit (0-9,*,#...). + + +#### asterisk/channel.h: fix documentation for 'ast_waitfor_nandfds()' + Author: Norm Harrison + Date: 2023-04-03 + + Co-authored-by: Florent CHAUVEAUBy Mark Spencer <markster@digium.com> and the Asterisk.org developer community.
Copyright (C) 2001-2025 Sangoma Technologies Corporation and other copyright holders.
@@ -37,7 +37,7 @@ hardware.
If you are updating from a previous version of Asterisk, make sure you
read the Change Logs.
-
+
NEW INSTALLATIONS
diff --git a/README.md b/README.md
index aa8057eca0..ce23ad5c34 100644
--- a/README.md
+++ b/README.md
@@ -55,7 +55,7 @@ If you are updating from a previous version of Asterisk, make sure you
read the Change Logs.
-[Change Logs](ChangeLogs/ChangeLog-21.8.0.html)
+[Change Logs](ChangeLogs/ChangeLog-21.9.0-rc1.html)
### NEW INSTALLATIONS