asterisk

Commit Graph

Author	SHA1	Message	Date
Sean Bright	541a907c12	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	dfa9cbc437	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	0f3e9fa565	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	6212c0b376	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	808c37333b	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
Asterisk Development Team	8d2135ec7a	Update for 20.17.0	3 months ago
Asterisk Development Team	ac4dcfdfbf	Update for 20.17.0-rc2	4 months ago
George Joseph	170b206925	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	4 months ago
Asterisk Development Team	1417052e49	Update for 20.17.0-rc1	4 months ago
Roman Pertsev	fe4fa6156f	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.	4 months ago
Sean Bright	a5c3280d60	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	4 months ago
George Joseph	db1a7d28bc	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.	4 months ago
Joshua C. Colp	fed0d484a5	Revert "pjsip: Move from threadpool to taskpool" This reverts commit `bb6b76c2d8`.	4 months ago
Tinet-mucw	dca0fa8bd4	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	4 months ago
George Joseph	98d0d2f33a	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	4 months ago
gauravs456	8c31864b55	chan_websocket: Add channel_id to MEDIA_START, DRIVER_STATUS and DTMF_END events. Resolves: #1544	4 months ago
Joshua C. Colp	da46b5f33f	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.	4 months ago
George Joseph	df4f6a7966	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	4 months ago
Christoph Moench-Tegeder	bced61f2be	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	4 months ago
Joshua C. Colp	029278046a	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.	4 months ago
George Joseph	07caf2ccda	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.	4 months ago
Sven Kube	14d82ecf62	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.	4 months ago
phoneben	d85047e44d	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	4 months ago
Joshua C. Colp	5dfb2f7200	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.	4 months ago
Joshua C. Colp	24bd5a6762	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.	4 months ago
George Joseph	411f66bd18	taskpool: Fix some references to threadpool that should be taskpool. Resolves: #1478	4 months ago
Anthony Minessale	ba9a4f7ab1	Update contact information for anthm	4 months ago
George Joseph	3feee6dfb6	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	4 months ago
Naveen Albert	d58e7ac759	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	4 months ago
Naveen Albert	599f0e0dfe	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.	4 months ago
Naveen Albert	51b1f2f7c7	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	4 months ago
Igor Goncharovsky	9ee4d452db	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.	4 months ago
Naveen Albert	cc768677c1	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	4 months ago
Naveen Albert	a2fcab4da9	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.	4 months ago
Naveen Albert	26d6e2489d	app_sf: Add post-digit timer option to ReceiveSF. Add a sorely needed option to set a timeout between digits, rather than for receiving the entire number. This is needed if the number of digits being sent is unknown by the receiver in advance. Previously, we had to wait for the entire timer to expire. Resolves: #1493 UserNote: The 't' option for ReceiveSF now allows for a timer since the last digit received, in addition to the number-wide timeout.	4 months ago
Naveen Albert	a2be1aad6b	res_tonedetect: Fix formatting of XML documentation. Fix the indentation in the documentation for the variable list. Resolves: #1507	4 months ago
Naveen Albert	e983fdc24f	res_fax: Add XML documentation for channel variables. Document the channel variables currently set by SendFAX and ReceiveFAX. Resolves: #1505	4 months ago
George Joseph	5002f9b7bd	channelstorage_cpp_map_name_id: Add read locking around retrievals. When we retrieve a channel from a C++ map, we actually get back a wrapper object that points to the channel then right after we retrieve it, we bump its reference count. There's a tiny chance however that between those two statements a delete and/or unref might happen which would cause the wrapper object or the channel itself to become invalid resulting in a SEGV. To avoid this we now perform a read lock on the driver around those statements. Resolves: #1491	4 months ago
Naveen Albert	41b2850562	app_dial: Allow fractional seconds for dial timeouts. Even though Dial() internally uses milliseconds for its dial timeouts, this capability has been mostly obscured from users as the argument is only parsed as an integer, thus forcing the use of whole seconds for timeouts. Parse it as a decimal instead so that timeouts can now truly have millisecond precision. Resolves: #1487 UserNote: The answer and progress dial timeouts now have millisecond precision, instead of having to be whole numbers.	4 months ago
Naveen Albert	8b5e6a7b3e	dsp.c: Make minor fixes to debug log messages. Commit `dc8e3eeaaf` improved the debug log messages in dsp.c. This makes two minor corrections to it: * Properly guard an added log statement in a conditional. * Don't add one to the hit count if there was no hit (however, we do still want to do this for the case where this is one). Resolves: #1496	4 months ago
Naveen Albert	38f8eb84b6	config_options.c: Improve misleading warning. When running "config show help <module>", if no XML documentation exists for the specified module, "Module <module> not found." is returned, which is misleading if the module is loaded but simply has no XML documentation for its config. Improve the message to clarify that the module may simply have no config documentation. Resolves: #1489	4 months ago
Naveen Albert	883cc61b67	func_scramble: Add example to XML documentation. The previous lack of an example made it ambiguous if the arguments went inside the function arguments or were part of the right-hand value. Resolves: #1485	4 months ago
Naveen Albert	86f1995622	sig_analog: Eliminate potential timeout with Last Number Redial. If Last Number Redial is used to redial, ensure that we do not wait for further digits. This was possible if the number that was last dialed is a prefix of another possible dialplan match. Since all we did is copy the number into the extension buffer, if other matches are now possible, there would thus be a timeout before the call went through. We now complete redialed calls immediaetly in all cases. Resolves: #1483	4 months ago
George Joseph	cfe7796af1	ARI: The bridges play and record APIs now handle sample rates > 8K correctly. The bridge play and record APIs were forcing the Announcer/Recorder channel to slin8 which meant that if you played or recorded audio with a sample rate > 8K, it was downsampled to 8K limiting the bandwidth. * The /bridges/play REST APIs have a new "announcer_format" parameter that allows the caller to explicitly set the format on the "Announcer" channel through which the audio is played into the bridge. If not specified, the default depends on how many channels are currently in the bridge. If a single channel is in the bridge, then the Announcer channel's format will be set to the same as that channel's. If multiple channels are in the bridge, the channels will be scanned to find the one with the highest sample rate and the Announcer channel's format will be set to the slin format that has an equal to or greater than sample rate. * The /bridges/record REST API has a new "recorder_format" parameter that allows the caller to explicitly set the format on the "Recorder" channel from which audio is retrieved to write to the file. If not specified, the Recorder channel's format will be set to the format that was requested to save the audio in. Resolves: #1479 DeveloperNote: The ARI /bridges/play and /bridges/record REST APIs have new parameters that allow the caller to specify the format to be used on the "Announcer" and "Recorder" channels respecitvely.	4 months ago
Max Grobecker	4281d3f3c3	res_pjsip_geolocation: Add support for Geolocation loc-src parameter This adds support for the Geolocation 'loc-src' parameter to res_pjsip_geolocation. The already existing config option 'location_source` in res_geolocation is documented to add a 'loc-src' parameter containing a user-defined FQDN to the 'Geolocation:' header, but that option had no effect as it was not implemented by res_pjsip_geolocation. If the `location_source` configuration option is not set or invalid, that parameter will not be added (this is already checked by res_geolocation). This commits adds already documented functionality.	4 months ago
Joshua C. Colp	45ea4176bb	sorcery: Move from threadpool to taskpool. This change moves observer invocation from the use of a threadpool to a taskpool. The taskpool options have also been adjusted to ensure that at least one taskprocessor remains available at all times.	4 months ago
Sven Kube	b2d5110a14	stasis_channels.c: Make protocol_id optional to enable blind transfer via ari When handling SIP transfers via ARI, there is no protocol_id in case of a blind transfer. Resolves: #1467	4 months ago
Bastian Triller	c2e03d8e8e	Fix some doxygen, typos and whitespace	4 months ago
Sven Kube	447b727ae1	stasis_channels.c: Add null check for referred_by in ast_ari_transfer_message_create When handling SIP transfers via ARI, the `referred_by` field in `transfer_ari_state` may be null, since SIP REFER requests are not required to include a `Referred-By` header. Without this check, a null value caused the transfer to fail and triggered a NOTIFY with a 500 Internal Server Error.	4 months ago
phoneben	2200871715	app_queue: Add NULL pointer checks in app_queue Add NULL check for word_list before calling word_in_list() Add NULL checks for channel snapshots from ast_multi_channel_blob_get_channel() Resolves: #1425	4 months ago

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