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