34276 Commits (d23fcd30a85b0fa0e32eff3e1c25ce2dba50b7d8)

 

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Author	SHA1	Message	Date
Joshua C. Colp	d23fcd30a8	pjsip: Move from threadpool to taskpool ... This change moves the PJSIP module from the threadpool API to the taskpool API. PJSIP-specific implementations for task usage have been removed and replaced with calls to the optimized taskpool implementations instead. The need for a pool of serializers has also been removed as taskpool inherently provides this. The default settings have also been changed to be more realistic for common usage. UpgradeNote: The threadpool_* options in pjsip.conf have now been deprecated though they continue to be read and used. They have been replaced with taskpool options that give greater control over the underlying taskpool used for PJSIP. An alembic upgrade script has been added to add these options to realtime as well.	2 weeks ago
phoneben	fbe735d613	Disable device state caching for ephemeral channels ... chan_audiosocket/chan_rtp/res_stasis_snoop: Disable device state caching for ephemeral channels Resolves: #1638	2 weeks ago
George Joseph	861b344295	chan_websocket: Add locking in send_event and check for NULL websocket handle. ... On an outbound websocket connection, when the triggering caller hangs up, webchan_hangup() closes the outbound websocket session and sets the websocket session handle to NULL. If the hangup happened in the tiny window between opening the outbound websocket connection and before read_thread_handler() was able to send the MEDIA_START message, it could segfault because the websocket session handle was NULL. If it didn't actually segfault, there was also the possibility that the websocket instance wouldn't get cleaned up which could also cause the channel snapshot to not get cleaned up. That could cause memory leaks and `core show channels` to list phantom WebSocket channels. To prevent the race, the send_event() macro now locks the websocket_pvt instance and checks the websocket session handle before attempting to send the MEDIA_START message. Resolves: #1643 Resolves: #1645	2 weeks ago
phoneben	9baa985119	Fix false null-deref warning in channel_state ... Resolve analyzer warning in channel_state by checking AST_FLAG_DEAD on snapshot, which is guaranteed non-NULL. Resolves: #1430	3 weeks ago
George Joseph	4c8381cb1e	endpoint.c: Plug a memory leak in ast_endpoint_shutdown(). ... Commit 26795be introduced a memory leak of ast_endpoint when ast_endpoint_shutdown() was called. The leak occurs only if a configuration change removes an endpoint and isn't related to call volume or the length of time asterisk has been running. An ao2_ref(-1) has been added to ast_endpoint_shutdown() to plug the leak. Resolves: #1635	3 weeks ago
Sean Bright	f255042167	Revert "func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()" ... This reverts commit 5177662990. For rationale, see #1621 and #1606	3 weeks ago
Sean Bright	1429addd29	cel_manager.c: Correct manager event mask for CEL events. ... There is no EVENT_FLAG_CEL and these events are raised with as EVENT_FLAG_CALL.	3 weeks ago
Sean Bright	5d6ba4d3e0	app_queue.c: Update docs to correct QueueMemberPause event name.	3 weeks ago
Mike Bradeen	8892d47409	taskprocessors: Improve logging and add new cli options ... This change makes some small changes to improve log readability in addition to the following changes: Modified 'core show taskprocessors' to now show Low time and High time for task execution. New command 'core show taskprocessor name <taskprocessor-name>' to dump taskprocessor info and current queue. Addionally, a new test was added to demonstrate the 'show taskprocessor name' functionality: test execute category /main/taskprocessor/ name taskprocessor_cli_show Setting 'core set debug 3 taskprocessor.c' will now log pushed tasks. (Warning this is will cause extremely high levels of logging at even low traffic levels.) Resolves: #1566 UserNote: New CLI command has been added - core show taskprocessor name <taskprocessor-name>	4 weeks ago
Michal Hajek	b5547a32bd	manager: fix double free of criteria variable when adding filter ... Signed-off-by: Michal Hajek <michal.hajek@daktela.com> Fixes: #1531	4 weeks ago
Sean Bright	8b0d7243ce	app_stream_echo.c: Check that stream is non-NULL before dereferencing. ... Also re-order and rename the arguments of `stream_echo_write_error` to match those of `ast_write_stream` for consistency. Resolves: #1427	4 weeks ago
Sean Bright	af4d21d291	abstract_jb.c: Remove redundant timer check per static analysis. ... While this check is technically unnecessary, it also was not harmful. The 2 other items mentioned in the linked issue are false positives and require no action. Resolves: #1417	4 weeks ago
phoneben	44d85b299d	channelstorage_cpp: Fix fallback return value in channelstorage callback ... callback returned the last iterated channel when no match existed, causing invalid channel references and potential double frees. Updated to correctly return NULL when there is no match. Resolves: #1609	4 weeks ago
George Joseph	bb6bf8a3f9	ccss: Add option to ccss.conf to globally disable it. ... The Call Completion Supplementary Service feature is rarely used but many of it's functions are called by app_dial and channel.c "just in case". These functions lock and unlock the channel just to see if CCSS is enabled on it, which it isn't 99.99% of the time. UserNote: A new "enabled" parameter has been added to ccss.conf. It defaults to "yes" to preserve backwards compatibility but CCSS is rarely used so setting "enabled = no" in the "general" section can save some unneeded channel locking operations and log message spam. Disabling ccss will also prevent the func_callcompletion and chan_dahdi modules from loading. DeveloperNote: A new API ast_is_cc_enabled() has been added. It should be used to ensure that CCSS is enabled before making any other ast_cc_* calls.	4 weeks ago
George Joseph	6156cdcb7d	app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE. ... UpgradeNote: In an effort to reduce log spam, two normal progress "pickup attempted" log messages from app_directed_pickup have been changed from NOTICE to VERBOSE(3). This puts them on par with other normal dialplan progress messages.	4 weeks ago
Sean Bright	e945b98d54	chan_websocket: Fix crash on DTMF_END event. ... Resolves: #1604	4 weeks ago
Joe Garlick	e5139be172	chan_websocket.c: Tolerate other frame types ... Currently, if chan_websocket receives an un supported frame like comfort noise it will exit the websocket. The proposed change is to tolerate the other frames by not sending them down the websocket but instead just ignoring them. Resolves: #1587	1 month ago
Naveen Albert	951ead13d8	app_reload: Fix Reload() without arguments. ... Calling Reload() without any arguments is supposed to reload everything (equivalent to a 'core reload'), but actually does nothing. This is because it was calling ast_module_reload with an empty string, and the argument needs to explicitly be NULL. Resolves: #1597	1 month ago
Naveen Albert	9306f6cdd0	pbx.c: Print new context count when reloading dialplan. ... When running "dialplan reload", the number of contexts reported is initially wrong, as it is the old context count. Running "dialplan reload" a second time returns the correct number of contexts that are loaded. This can confuse users into thinking that the reload didn't work successfully the first time. This counter is currently only incremented when iterating the old contexts prior to the context merge; at the very end, get the current number of elements in the context hash table and report that instead. This way, the count is correct immediately whenever a reload occurs. Resolves: #1599	1 month ago
C. Maj	41f7d0d320	Makefile: Add module-list-* targets. ... Convenience wrappers for showing modules at various support levels. * module-list-core * module-list-extended * module-list-deprecated Resolves: #1572 UserNote: Try "make module-list-deprecated" to see what modules are on their way out the door.	1 month ago
Naveen Albert	66f337ac51	app_disa: Avoid use of removed ResetCDR() option. ... Commit a46d5f9b76 removed the deprecated 'e' option to ResetCDR; this now causes DISA() to emit a warning if attempting to call ResetCDR() with the deprecated option (in all cases except when the no answer option is provided). Rewrite the code to do this the current way. Resolves: #1592	1 month ago
Tinet-mucw	f8597e0991	core_unreal.c: Use ast instead of p->chan to get the DIALSTATUS variable ... After p->chan = NULL, ast still points to the valid channel object, using ast safely accesses the channel's DIALSTATUS variable before it's fully destroyed Resolves: #1590	1 month ago
George Joseph	9bb18e0e36	ast_coredumper: Fix multiple issues ... * Fixed an issue with tarball-coredumps when asterisk was invoked without an absolute path. * Fixed an issue with gdb itself segfaulting when trying to get symbols from separate debuginfo files. The command line arguments needed to be altered such that the gdbinit files is loaded before anything else but the `dump-asterisk` command is run after full initialization. In the embedded gdbinit script: * The extract_string_symbol function needed a `char *` cast to work properly. * The s_strip function needed to be updated to continue to work with the cpp_map_name_id channel storage backend. * A new function was added to dump the channels when cpp_map_name_id was used. * The Channel object was updated to account for the new channel storage backends * The show_locks function was refactored to work correctly.	2 months ago
Daouda Taha	2e699dd95c	app_mixmonitor: Add 's' (skip) option to delay recording. ... The 's' (skip) option delays MixMonitor recording until the specified number of seconds (can be fractional) have elapsed since MixMonitor was invoked. No audio is written to the recording file during this time. If the call ends before this period, no audio will be saved. This is useful for avoiding early audio such as announcements, ringback tones, or other non-essential sounds. UserNote: This change introduces a new 's(<seconds>)' (skip) option to the MixMonitor application. Example: MixMonitor(${UNIQUEID}.wav,s(3)) This skips recording for the first 3 seconds before writing audio to the file. Existing MixMonitor behavior remains unchanged when the 's' option is not used.	2 months ago
phoneben	82a965f571	stasis: switch stasis show topics temporary container from list - RBtree ... switch stasis show topics temporary container from list to RB-tree minimizing lock time Resolves: #1585	2 months ago
Sean Bright	4a7c94a9cc	app_dtmfstore: Avoid a potential buffer overflow. ... Prefer snprintf() so we can readily detect if our output was truncated. Resolves: #1421	2 months ago
Sean Bright	af1bb0f33e	main: Explicitly mark case statement fallthrough as such. ... Resolves: #1442	2 months ago
Sean Bright	9f97dfaa44	bridge_softmix: Return early on topology allocation failure. ... Resolves: #1446	2 months ago
Sean Bright	3d0f10e1e8	bridge_simple: Increase code verbosity for clarity. ... There's no actual problem here, but I can see how it might by confusing. Resolves: #1444	2 months ago
Kristian F. Høgh	092147cc8c	app_queue.c: Only announce to head caller if announce_to_first_user ... Only make announcements to head caller if announce_to_first_user is true Fixes: #1568 UserNote: When announce_to_first_user is false, no announcements are played to the head caller	2 months ago
George Joseph	7aa0b6434e	channelstorage: Allow storage driver read locking to be skipped. ... After PR #1498 added read locking to channelstorage_cpp_map_name_id, if ARI channels/externalMedia was called with a custom channel id AND the cpp_map_name_id channel storage backend is in use, a deadlock can occur when hanging up the channel. It's actually triggered in channel.c:__ast_channel_alloc_ap() when it gets a write lock on the channelstorage driver then subsequently does a lookup for channel uniqueid which now does a read lock. This is an invalid operation and causes the lock state to get "bad". When the channels try to hang up, a write lock is attempted again which hangs and causes the deadlock. Now instead of the cpp_map_name_id channelstorage driver "get" APIs automatically performing a read lock, they take a "lock" parameter which allows a caller who already has a write lock to indicate that the "get" API must not attempt its own lock. This prevents the state from getting mesed up. The ao2_legacy driver uses the ao2 container's recursive mutex so doesn't have this issue but since it also implements the common channelstorage API, it needed its "get" implementations updated to take the lock parameter. They just don't use it. Resolves: #1578	2 months ago
George Joseph	7813090e4f	chan_websocket: Add ability to place a MARK in the media stream. ... Also cleaned up a few unused #if blocks, and started sending a few ERROR events back to the apps. Resolves: #1574 DeveloperNote: Apps can now send a `MARK_MEDIA` command with an optional `correlation_id` parameter to chan_websocket which will be placed in the media frame queue. When that frame is dequeued after all intervening media has been played to the core, chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the app with the same correlation_id (if any).	2 months ago
George Joseph	650c68e306	chan_websocket: Add capability for JSON control messages and events. ... With recent enhancements to chan_websocket, the original plain-text implementation of control messages and events is now too limiting. We probably should have used JSON initially but better late than never. Going forward, enhancements that require control message or event changes will only be done to the JSON variants and the plain-text variants are now deprecated but not yet removed. * Added the chan_websocket.conf config file that allows setting which control message format to use globally: "json" or "plain-text". "plain-text" is the default for now to preserve existing behavior. * Added a dialstring option `f(json|plain-text)` to allow the format to be overridden on a call-by-call basis. Again, 'plain-text' is the default for now to preserve existing behavior. The JSON for commands sent by the app to Asterisk must be... `{ "command": "<command>" ... }` where `<command>` is one of `ANSWER`, `HANGUP`, `START_MEDIA_BUFFERING`, etc. The `STOP_MEDIA_BUFFERING` command takes an additional, optional parameter to be returned in the corresponding `MEDIA_BUFFERING_COMPLETED` event: `{ "command": "STOP_MEDIA_BUFFERING", "correlation_id": "<correlation id>" }`. The JSON for events sent from Asterisk to the app will be... `{ "event": "<event>", "channel_id": "<channel_id>" ... }`. The `MEDIA_START` event will now look like... ``` { "event": "MEDIA_START", "connection_id": "media_connection1", "channel": "WebSocket/media_connection1/0x5140001a0040", "channel_id": "1761245643.1", "format": "ulaw", "optimal_frame_size": 160, "ptime": 20, "channel_variables": { "DIALEDPEERNUMBER": "media_connection1/c(ulaw)", "MEDIA_WEBSOCKET_CONNECTION_ID": "media_connection1", "MEDIA_WEBSOCKET_OPTIMAL_FRAME_SIZE": "160" } } ``` Note the addition of the channel variables which can't be supported with the plain-text formatting. The documentation will be updated with the exact formats for all commands and events. Resolves: #1546 Resolves: #1563 DeveloperNote: The chan_websocket plain-text control and event messages are now deprecated (but remain the default) in favor of JSON formatted messages. See https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket for more information. DeveloperNote: A "transport_data" parameter has been added to the channels/externalMedia ARI endpoint which, for websocket, allows the caller to specify parameters to be added to the dialstring for the channel. For instance, `"transport_data": "f(json)"`.	2 months ago
George Joseph	d510b85b45	build: Add menuselect options to facilitate code tracing and coverage ... The following options have been added to the menuselect "Compiler Flags" section... CODE_COVERAGE: The ability to enable code coverage via the `--enable-coverage` configure flag has existed for many years but changing it requires re-running ./configure which is painfully slow. With this commit, you can now enable and disable it via menuselect. Setting this option adds the `-ftest-coverage` and `-fprofile-arcs` flags on the gcc and ld command lines. It also sets DONT_OPTIMIZE. Note: If you use the `--enable-coverage` configure flag, you can't turn it off via menuselect so choose one method and stick to it. KEEP_FRAME_POINTERS: This option sets `-fno-omit-frame-pointers` on the gcc command line which can facilitate debugging with 'gdb' and tracing with 'perf'. Unlike CODE_COVERAGE, this option doesn't depend on optimization being disabled. It does however conflict with COMPILE_DOUBLE.	2 months ago
Roman Pertsev	797902effc	res_audiosocket: fix temporarily unavailable ... Operations on non-blocking sockets may return a resource temporarily unavailable error (EAGAIN or EWOULDBLOCK). This is not a fatal error but a normal condition indicating that the operation would block. This patch corrects the handling of this case. Instead of incorrectly treating it as a reason to terminate the connection, the code now waits for data to arrive on the socket.	2 months ago
Sean Bright	b9c71dd40d	safe_asterisk: Resolve a POSIX sh problem and restore globbing behavior. ... * Using `==` with the POSIX sh `test` utility is UB. * Switch back to using globs instead of using `$(find … | sort)`. * Fix a missing redirect when checking for the OS type. Resolves: #1554	2 months ago
George Joseph	086f1893ec	res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function. ... Also... * Refactored the verification datastore process so instead of having a separate channel datastore for each verification result, there's only one channel datastore with a vector of results. * Refactored some log messages to include channel name and removed some that would be redundant if a memory allocation failed. Resolves: #781 UserNote: The STIR_SHAKEN_ATTESTATION dialplan function has been added which will allow suppressing attestation on a call-by-call basis regardless of the profile attached to the outgoing endpoint.	2 months ago
Joshua C. Colp	b2e17cb78d	Revert "pjsip: Move from threadpool to taskpool" ... This reverts commit bb6b76c2d8.	2 months ago
Tinet-mucw	3528bedd56	iostream.c: Handle TLS handshake attacks in order to resolve the issue of exceeding the maximum number of HTTPS sessions. ... The TCP three-way handshake completes, but if the server is under a TLS handshake attack, asterisk will get stuck at SSL_do_handshake(). In this case, a timeout mechanism should be set for the SSL/TLS handshake process to prevent indefinite waiting during the SSL handshake. Resolves: #1559	2 months ago
George Joseph	8e8a0a1e82	chan_pjsip: Disable SSRC change for WebRTC endpoints. ... Commit b333ee3b introduced a fix to chan_pjsip that addressed RTP issues with blind transfers and some SBCs. Unfortunately, the fix broke some WebRTC clients that are sensitive to SSRC changes and non-monotonic timestamps so the fix is now disabled for endpoints with the "bundle" parameter set to true. Resolves: #1535	2 months ago
gauravs456	0019b27f5c	chan_websocket: Add channel_id to MEDIA_START, DRIVER_STATUS and DTMF_END events. ... Resolves: #1544	2 months ago
Joshua C. Colp	2d28889193	pjsip: Move from threadpool to taskpool ... This change moves the PJSIP module from the threadpool API to the taskpool API. PJSIP-specific implementations for task usage have been removed and replaced with calls to the optimized taskpool implementations instead. The need for a pool of serializers has also been removed as taskpool inherently provides this. The default settings have also been changed to be more realistic for common usage. UpgradeNote: The threadpool_* options in pjsip.conf have now been deprecated though they continue to be read and used. They have been replaced with taskpool options that give greater control over the underlying taskpool used for PJSIP. An alembic upgrade script has been added to add these options to realtime as well.	2 months ago
George Joseph	74a3809be8	safe_asterisk: Fix logging and sorting issue. ... Re-enabled "TTY=9" which was erroneously disabled as part of a recent security fix and removed another logging "fix" that was added. Also added a sort to the "find" that enumerates the scripts to be sourced so they're sourced in the correct order. Resolves: #1539	2 months ago
Christoph Moench-Tegeder	dc55f3f372	Fix Endianness detection in utils.h for non-Linux ... Commit 43bf8a4ded introduced endian dependend byte-swapping code in include/asterisk/utils.h, where the endianness was detected using the __BYTE_ORDER macro. This macro lives in endian.h, which on Linux is included implicitely (by the network-related headers, I think), but on FreeBSD the headers are laid out differently and we do not get __BYTE_ORDER the implicit way. Instead, this makes the usage of endian.h explicit by including it where we need it, and switches the BYTE_ORDER/*ENDIAN macros to the POSIX-defined ones (see https://pubs.opengroup.org/onlinepubs/9799919799/basedefs/endian.h.html for standard compliance). Additionally, this adds a compile-time check for the endianness-logic: compilation will fail if neither big nor little endian can be detected. Fixes: #1536	2 months ago
Ben Ford	d5e88e7ed9	app_queue.c: Fix error in Queue parameter documentation. ... When macro was removed in Asterisk 21, the parameter documentation in code was not updated to reflect the correct numerization for gosub. It still stated that it was the seventh parameter, but got shifted to the sixth due to the removal of macro. This has been updated to correctly reflect the parameter order, and a note has been added to the XML that states this was done after the initial commit. Fixes: #1534 UpgradeNote: As part of Asterisk 21, macros were removed from Asterisk. This resulted in argument order changing for the Queue dialplan application since the macro argument was removed. Upgrade notice was missed when this was done, so this upgrade note has been added to provide a record of such and a notice to users who may have not upgraded yet.	2 months ago
Joshua C. Colp	ccc5b584f4	devicestate: Don't publish redundant device state messages. ... When publishing device state check the local cache for the existing device state. If the new device state is unchanged from the prior one, don't bother publishing the update. This can reduce the work done by consumers of device state, such as hints and app_queue, by not publishing a message to them. These messages would most often occur with devices that are seeing numerous simultaneous channels. The underlying device state would remain as in use throughout, but an update would be published as channels are created and hung up.	2 months ago
George Joseph	683f3fb387	chan_pjsip: Add technology-specific off-nominal hangup cause to events. ... Although the ISDN/Q.850/Q.931 hangup cause code is already part of the ARI and AMI hangup and channel destroyed events, it can be helpful to know what the actual channel technology code was if the call was unsuccessful. For PJSIP, it's the SIP response code. * A new "tech_hangupcause" field was added to the ast_channel structure along with ast_channel_tech_hangupcause() and ast_channel_tech_hangupcause_set() functions. It should only be set for off-nominal terminations. * chan_pjsip was modified to set the tech hangup cause in the chan_pjsip_hangup() and chan_pjsip_session_end() functions. This is a bit tricky because these two functions aren't always called in the same order. The channel that hangs up first will get chan_pjsip_session_end() called first which will trigger the core to call chan_pjsip_hangup() on itself, then call chan_pjsip_hangup() on the other channel. The other channel's chan_pjsip_session_end() function will get called last. Unfortunately, the other channel's HangupRequest events are sent before chan_pjsip has had a chance to set the tech hangupcause code so the HangupRequest events for that channel won't have the cause code set. The ChannelDestroyed and Hangup events however will have the code set for both channels. * A new "tech_cause" field was added to the ast_channel_snapshot_hangup structure. This is a public structure so a bit of refactoring was needed to preserve ABI compatibility. * The ARI ChannelHangupRequest and ChannelDestroyed events were modified to include the "tech_cause" parameter in the JSON for off-nominal terminations. The parameter is suppressed for nominal termination. * The AMI SoftHangupRequest, HangupRequest and Hangup events were modified to include the "TechCause" parameter for off-nominal terminations. Like their ARI counterparts, the parameter is suppressed for nominal termination. DeveloperNote: A "tech_cause" parameter has been added to the ChannelHangupRequest and ChannelDestroyed ARI event messages and a "TechCause" parameter has been added to the HangupRequest, SoftHangupRequest and Hangup AMI event messages. For chan_pjsip, these will be set to the last SIP response status code for off-nominally terminated calls. The parameter is suppressed for nominal termination.	2 months ago
Sven Kube	cb5b22db1f	res_audiosocket: add message types for all slin sample rates ... Extend audiosocket messages with types 0x11 - 0x18 to create asterisk frames in slin12, slin16, slin24, slin32, slin44, slin48, slin96, and slin192 format, enabling the transmission of audio at a higher sample rates. For audiosocket messages sent by Asterisk, the message kind is determined by the format of the originating asterisk frame. UpgradeNote: New audiosocket message types 0x11 - 0x18 has been added for slin12, slin16, slin24, slin32, slin44, slin48, slin96, and slin192 audio. External applications using audiosocket may need to be updated to support these message types if the audiosocket channel is created with one of these audio formats.	2 months ago
phoneben	7cc1eb3036	res_fax.c: lower FAXOPT read warning to debug level ... Reading ${FAXOPT()} before a fax session is common in dialplans to check fax state. Currently this logs an error even when no fax datastore exists, creating excessive noise. Change these messages to ast_debug(3, …) so they appear only with debug enabled. Resolves: #1509	3 months ago
Joshua C. Colp	7477d120fd	endpoints: Remove need for stasis subscription. ... When an endpoint is created in the core of Asterisk a subscription was previously created alongside it to monitor any channels being destroyed that were related to it. This was done by receiving all channel snapshot updates for every channel and only reacting when it was indicated that the channel was dead. This change removes this logic and instead provides an API call for directly removing a channel from an endpoint. This is called when channels are destroyed. This operation is fast, so blocking the calling thread for a short period of time doesn't have any noticeable impact.	3 months ago