asterisk

Commit Graph

Author	SHA1	Message	Date
George Joseph	7c9133c854	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.	8 months ago
George Joseph	193b580f7d	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.	8 months ago
Sean Bright	e5b15f0206	chan_websocket: Fix crash on DTMF_END event. Resolves: #1604	8 months ago
Joe Garlick	7d6b43cd99	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	8 months ago
Naveen Albert	c6daea7052	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	8 months ago
Naveen Albert	5354e8a001	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	8 months ago
C. Maj	b762e47863	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.	8 months ago
Naveen Albert	f401f1c1fc	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	8 months ago
Tinet-mucw	396a4f153f	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	8 months ago
George Joseph	a90af28ccf	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.	8 months ago
Daouda Taha	c182e91ead	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.	8 months ago
phoneben	390ca4e827	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	8 months ago
Sean Bright	b9141cb4be	app_dtmfstore: Avoid a potential buffer overflow. Prefer snprintf() so we can readily detect if our output was truncated. Resolves: #1421	8 months ago
Sean Bright	3238e4e495	main: Explicitly mark case statement fallthrough as such. Resolves: #1442	8 months ago
Sean Bright	6f920eeeae	bridge_softmix: Return early on topology allocation failure. Resolves: #1446	8 months ago
Sean Bright	02f3f341d1	bridge_simple: Increase code verbosity for clarity. There's no actual problem here, but I can see how it might by confusing. Resolves: #1444	8 months ago
Kristian F. Høgh	6e631a5c1c	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	8 months ago
George Joseph	98e8f081d4	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).	8 months ago
George Joseph	27ed121f27	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)"`.	8 months ago
George Joseph	57b409bdf8	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.	8 months ago
Asterisk Development Team	7b98cadd19	Update for 22.7.0	10 months ago
Asterisk Development Team	9b26c4370e	Update for 22.7.0-rc2	11 months ago
George Joseph	2c75447235	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
Asterisk Development Team	2c73433fb8	Update for 22.7.0-rc1	11 months ago
Roman Pertsev	481d4c0e75	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	7e5b531828	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	94d04ac311	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	54da3e1b9e	Revert "pjsip: Move from threadpool to taskpool" This reverts commit `bb6b76c2d8`.	11 months ago
Tinet-mucw	38f2fd2527	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
George Joseph	08960ea5a1	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	11 months ago
gauravs456	b03582c30e	chan_websocket: Add channel_id to MEDIA_START, DRIVER_STATUS and DTMF_END events. Resolves: #1544	11 months ago
Joshua C. Colp	7662c5cb6c	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.	11 months ago
George Joseph	c3b4b923b3	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	11 months ago
Christoph Moench-Tegeder	9bbb566df5	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	11 months ago
Ben Ford	b1a14698a7	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.	11 months ago
Joshua C. Colp	b6556cc789	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.	11 months ago
George Joseph	1b98cd1b25	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.	11 months ago
Sven Kube	da10edda02	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.	11 months ago
phoneben	316b4731cb	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	11 months ago
Joshua C. Colp	5b4aac6ba0	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.	11 months ago
Joshua C. Colp	3202f8c16d	app_queue: Allow stasis message filtering to work. The app_queue module subscribes on a per-dialed agent basis to both the bridge all and channel all topics to keep apprised of things going on involving them. This subscription has associated state that must be cleaned up when the subscription ends. This was done by setting a default router callback that only had logic to handle the case where the subscription ends. By using the default router callback all filtering for the subscription was disabled, causing unrelated messages to get published and handled by it. This change makes it so that an explicit route is added for the message type used for the message indicating the subscription has ended and removes the default router callback. This allows message filtering to occur on publishing reducing the messages to app_queue to only those it is interested in.	11 months ago
George Joseph	b3e5ca59a0	taskpool: Fix some references to threadpool that should be taskpool. Resolves: #1478	11 months ago
Anthony Minessale	76ca632967	Update contact information for anthm	11 months ago
George Joseph	edcd8b4968	chan_websocket.c: Change payload references to command instead. Some of the tests in process_text_message() were still comparing to the websocket message payload instead of the "command" string. Resolves: #1525	11 months ago
Naveen Albert	a58ddff893	func_callerid: Document limitation of DNID fields. The Dial() application does not propagate DNID fields, which is counter to the behavior of the other Caller ID fields. This behavior is likely intentional since the use of Dial theoretically suggests a new dialed number, but document this caveat to inform users of it. Resolves: #1519	11 months ago
Naveen Albert	241d07f6b4	func_channel: Allow R/W of ADSI CPE capability setting. Allow retrieving and setting the channel's ADSI capability from the dialplan. Resolves: #1514 UserNote: CHANNEL(adsicpe) can now be read or written to change the channels' ADSI CPE capability setting.	11 months ago
Naveen Albert	9a4f6f4c4a	core_unreal: Preserve ADSI capability when dialing Local channels. Dial() already preserves the ADSI capability by copying it to the new channel, but since Local channel pairs consist of two channels, we also need to copy the capability to the second channel. Resolves: #1517	11 months ago
Igor Goncharovsky	17ae831db5	func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE() As soon as SIP call may end with several Reason headers, we want to make all of them available through the HAGUPCAUSE() function. This implementation uses the same ao2 hash for cause codes storage and adds a flag to make difference between last processed sip message and content of reason headers. UserNote: Added a new option to HANGUPCAUSE to access additional information about hangup reason. Reason headers from pjsip could be read using 'tech_extended' cause type.	11 months ago
Naveen Albert	9b96b6bd59	sig_analog: Allow '#' to end the inter-digit timeout when dialing. It is customary to allow # to terminate digit collection immediately when there would normally be a timeout. However, currently, users are forced to wait for the timeout to expire when dialing numbers that are prefixes of other valid matches, and there is no way to end the timeout early. Customarily, # terminates the timeout, but at the moment, this is just rejected unless there happens to be a matching extension ending in #. Allow # to terminate the timeout in cases where there is no dialplan match. This ensures that the dialplan is always respected, but if a valid extension has been dialed that happens to prefix other valid matches, # can be used to dial it immediately. Resolves: #1510	11 months ago
Naveen Albert	ff40d2eb04	func_math: Add DIGIT_SUM function. Add a function (DIGIT_SUM) which returns the digit sum of a number. Resolves: #1499 UserNote: The DIGIT_SUM function can be used to return the digit sum of a number.	11 months ago

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