34303 Commits (d8caa664a8e1be4b9b85c7a0eac17ba498dbcc6a)

 

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Author	SHA1	Message	Date
Maximilian Fridrich	d8caa664a8	res_pjsip_messaging: Add support for following 3xx redirects ... This commit integrates the redirect module into res_pjsip_messaging to enable following 3xx redirect responses for outgoing SIP MESSAGEs. When follow_redirect_methods contains 'message' on an endpoint, Asterisk will now follow 3xx redirect responses for MESSAGEs, similar to how it behaves for INVITE responses. Resolves: #1576 UserNote: A new pjsip endpoint option follow_redirect_methods was added. This option is a comma-delimited, case-insensitive list of SIP methods for which SIP 3XX redirect responses are followed. An alembic upgrade script has been added for adding this new option to the Asterisk database.	9 months ago
Maximilian Fridrich	2bd999ae5f	res_pjsip: Introduce redirect module for handling 3xx responses ... This commit introduces a new redirect handling module that provides infrastructure for following SIP 3xx redirect responses. The redirect functionality respects the endpoint's redirect_method setting and only follows redirects when set to 'uri_pjsip'. This infrastructure can be used by any PJSIP module that needs to handle 3xx redirect responses.	9 months ago
Tinet-mucw	f91f7d580d	app_mixmonitor.c: Fix crash in mixmonitor_ds_remove_and_free when datastore is NULL ... The datastore may be NULL, so a null pointer check needs to be added. Resolves: #1673	9 months ago
Sven Kube	55e403fa8b	res_pjsip_refer: don't defer session termination for ari transfer ... Allow session termination during an in progress ari handled transfer.	9 months ago
Naveen Albert	18da95e8a8	chan_dahdi.conf.sample: Avoid warnings with default configs. ... callgroup and pickupgroup may only be specified for FXO-signaled channels; however, the chan_dahdi sample config had these options uncommented in the [channels] section, thus applying these settings to all channels, resulting in warnings. Comment these out so there are no warnings with an unmodified sample config. Resolves: #1552	9 months ago
sarangr7	3b4f4d2258	main/dial.c: Set channel hangup cause on timeout in handle_timeout_trip ... When dial attempts timeout in the core dialing API, the channel's hangup cause was not being set before hanging up. Only the ast_dial_channel structure's internal cause field was updated, but the actual ast_channel hangup cause remained unset. This resulted in incorrect or missing hangup cause information being reported through CDRs, AMI events, and other mechanisms that read the channel's hangup cause when dial timeouts occurred via applications using the dialing API (FollowMe, Page, etc.). The fix adds proper channel locking and sets AST_CAUSE_NO_ANSWER on the channel before calling ast_hangup(), ensuring consistent hangup cause reporting across all interfaces. Resolves: #1660	9 months ago
Sean Bright	a9b11243eb	cel: Add missing manager documentation. ... The LOCAL_OPTIMIZE_BEGIN, STREAM_BEGIN, STREAM_END, and DTMF CEL events were not all documented in the CEL configuration file or the manager documentation for the CEL event.	9 months ago
Sean Bright	719c882358	res_odbc: Use SQL_SUCCEEDED() macro where applicable. ... This is just a cleanup of some repetitive code.	9 months ago
Justin T. Gibbs	ce0eca443f	rtp/rtcp: Configure dual-stack behavior via IPV6_V6ONLY ... Dual-stack behavior (simultaneous listening for IPV4 and IPV6 connections on a single socket) is required by Asterisk's ICE implementation. On systems with the IPV6_V6ONLY sockopt, set the option to 0 (dual-stack enabled) when binding to the IPV6 any address. This ensures correct behavior regardless of the system's default dual-stack configuration.	9 months ago
Sean Bright	08d4960df7	http.c: Include remote address in URI handler message. ... Resolves: #1662	9 months ago
Joshua C. Colp	c67d3d2796	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.	9 months ago
phoneben	81f73cb4d8	Disable device state caching for ephemeral channels ... chan_audiosocket/chan_rtp/res_stasis_snoop: Disable device state caching for ephemeral channels Resolves: #1638	9 months ago
George Joseph	1f77cdc7c3	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	9 months ago
phoneben	20e40a9a48	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	10 months ago
George Joseph	bf0ce4577c	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	10 months ago
Sean Bright	3e84ffaaa1	Revert "func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()" ... This reverts commit 5177662990. For rationale, see #1621 and #1606	10 months ago
Paul Donald	c798d3a3ff	configs: rename phoneprov_users.conf to .sample so make installs it ... This conf file should be suffixed .sample so that make installs it at compile time. Otherwise res_phoneprov complains at runtime as to its absence and refuses to start. Fixes: #1626	10 months ago
Sean Bright	0d2f200bfb	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.	10 months ago
Sean Bright	3ce5308ac9	app_queue.c: Update docs to correct QueueMemberPause event name.	10 months ago
Mike Bradeen	f29ce05e3e	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>	10 months ago
Michal Hajek	0cf61f8d47	manager: fix double free of criteria variable when adding filter ... Signed-off-by: Michal Hajek <michal.hajek@daktela.com> Fixes: #1531	10 months ago
Sean Bright	9647a4f022	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	10 months ago
Sean Bright	2ecdc84d07	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	10 months ago
phoneben	065c507ffc	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	10 months ago
George Joseph	ed8d5c6430	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.	10 months ago
George Joseph	23b5de4923	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.	10 months ago
Sean Bright	ed46dd0e36	chan_websocket: Fix crash on DTMF_END event. ... Resolves: #1604	10 months ago
Joe Garlick	5d15a42af1	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	10 months ago
Naveen Albert	e38a5b7c87	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	10 months ago
Naveen Albert	b19dbffa9a	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	10 months ago
C. Maj	183732d172	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.	10 months ago
Naveen Albert	ba9d9ec64a	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	10 months ago
Tinet-mucw	fe194b4016	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	10 months ago
George Joseph	bb506faaa7	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.	11 months ago
Daouda Taha	7d104d69a9	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.	11 months ago
phoneben	f9e00d3ead	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	11 months ago
Sean Bright	61b2538e9b	app_dtmfstore: Avoid a potential buffer overflow. ... Prefer snprintf() so we can readily detect if our output was truncated. Resolves: #1421	11 months ago
Sean Bright	86c3baea61	main: Explicitly mark case statement fallthrough as such. ... Resolves: #1442	11 months ago
Sean Bright	c520bd45a1	bridge_softmix: Return early on topology allocation failure. ... Resolves: #1446	11 months ago
Sean Bright	33987ed966	bridge_simple: Increase code verbosity for clarity. ... There's no actual problem here, but I can see how it might by confusing. Resolves: #1444	11 months ago
Kristian F. Høgh	1a594fcb21	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	11 months ago
George Joseph	c1e5258497	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	11 months ago
George Joseph	4fa314242b	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).	11 months ago
George Joseph	f08020df80	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)"`.	11 months ago
George Joseph	a9a1b976a7	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.	11 months ago
Roman Pertsev	27fb039839	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.	11 months ago
Sean Bright	1564548980	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	11 months ago
George Joseph	438649b30d	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.	11 months ago
Joshua C. Colp	74b1cafece	Revert "pjsip: Move from threadpool to taskpool" ... This reverts commit bb6b76c2d8.	11 months ago
Tinet-mucw	b026fb80bd	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	11 months ago