asterisk

Commit Graph

Author	SHA1	Message	Date
George Joseph	487bf8af26	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	88f10346ec	Update for 23.1.0-rc1	11 months ago
Roman Pertsev	920712396b	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	8a73d4939d	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	8f459f783f	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	3581473920	Revert "pjsip: Move from threadpool to taskpool" This reverts commit `bb6b76c2d8`.	11 months ago
Tinet-mucw	13c5047ff8	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	c98ac2c6f3	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	c521337a1f	chan_websocket: Add channel_id to MEDIA_START, DRIVER_STATUS and DTMF_END events. Resolves: #1544	11 months ago
Joshua C. Colp	3d21ae062d	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	0610a348eb	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	64e2eaaf0f	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	f88977d03e	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	7de8460ba6	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	ce9c44292a	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	845c292e1b	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	8bf568906b	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	a624208b32	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	324e69f3d5	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	d8d79a6bf7	taskpool: Fix some references to threadpool that should be taskpool. Resolves: #1478	11 months ago
Anthony Minessale	fab6098cc9	Update contact information for anthm	11 months ago
George Joseph	0c0ceb71d5	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	d1872278c6	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	576c559d32	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	834e8923d4	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	15b4e46063	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	309afd3106	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	46c079b9db	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
Naveen Albert	d0c2d0394b	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.	11 months ago
Naveen Albert	22efe33457	codec_builtin.c: Adjust some of the quality scores to reflect reality. Among the lower-quality voice codecs, some of the quality scores did not make sense relative to each other. For instance, quality-wise, G.729 > G.723 > PLC10. However, current scores do not uphold these relationships. Tweak the scores slightly to reflect more accurate relationships. Resolves: #1501	11 months ago
Naveen Albert	d9685441c1	res_tonedetect: Fix formatting of XML documentation. Fix the indentation in the documentation for the variable list. Resolves: #1507	11 months ago
Naveen Albert	2f166aec15	res_fax: Add XML documentation for channel variables. Document the channel variables currently set by SendFAX and ReceiveFAX. Resolves: #1505	11 months ago
George Joseph	3719246c40	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	11 months ago
Naveen Albert	a220c5f154	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.	11 months ago
Naveen Albert	75b3832748	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	11 months ago
Naveen Albert	5b936db6e5	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	11 months ago
Naveen Albert	ff854114b1	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	11 months ago
Naveen Albert	3bd3f04cfa	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	11 months ago
George Joseph	f5dea21b91	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.	11 months ago
Max Grobecker	6cf498ba43	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.	11 months ago
Joshua C. Colp	9952507dfb	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.	11 months ago
Sven Kube	d48e370811	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	11 months ago
Allan Nathanson	3781493e02	config.c: fix saving of deep/wide template configurations Follow-on to #244 and #960 regarding how the ast_config_XXX APIs handle template inheritance. ast_config_text_file_save2() incorrectly suppressed variables if they matched any ancestor template. This broke deep chains (dropping values based on distant parents) and wide inheritance (ignoring last-wins order across multiple parents). The function now inspects the full template hierarchy to find the nearest effective parent (last occurrence wins). Earlier inherited duplicates are collapsed, explicit overrides are kept unless they exactly match the parent, and PreserveEffectiveContext avoids writing redundant lines. Resolves: #1451	11 months ago
Bastian Triller	b9847b70c5	Fix some doxygen, typos and whitespace	11 months ago
Sven Kube	884c228280	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.	11 months ago
phoneben	b502b7cf55	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	11 months ago
Sean Bright	a4d638fa02	app_externalivr: Prevent out-of-bounds read during argument processing. Resolves: #1422	11 months ago
Naveen Albert	930f5578e5	chan_dahdi: Add DAHDI_CHANNEL function. Add a dialplan function that can be used to get/set properties of DAHDI channels (as opposed to Asterisk channels). This exposes properties that were not previously available, allowing for certain operations to now be performed in the dialplan. Resolves: #1455 UserNote: The DAHDI_CHANNEL function allows for getting/setting certain properties about DAHDI channels from the dialplan.	11 months ago
Joshua C. Colp	b5729635d5	taskpool: Update versions for taskpool stasis options.	11 months ago
Joshua C. Colp	37800188ec	taskpool: Add taskpool API, switch Stasis to using it. This change introduces a new API called taskpool. This is a pool of taskprocessors. It provides the following functionality: 1. Task pushing to a pool of taskprocessors 2. Synchronous tasks 3. Serializers for execution ordering of tasks 4. Growing/shrinking of number of taskprocessors in pool This functionality already exists through the combination of threadpool+taskprocessors but through investigating I determined that this carries substantial overhead for short to medium duration tasks. The threadpool uses a single queue of work, and for management of threads it involves additional tasks. I wrote taskpool to eliminate the extra overhead and management as much as possible. Instead of a single queue of work each taskprocessor has its own queue and at push time a selector chooses the taskprocessor to queue the task to. Each taskprocessor also has its own thread like normal. This spreads out the tasks immediately and reduces contention on shared resources. Using the included efficiency tests the number of tasks that can be executed per second in a taskpool is 6-12 times more than an equivalent threadpool+taskprocessor setup. Stasis has been moved over to using this new API as it is a heavy consumer of threadpool+taskprocessors and produces a lot of tasks. UpgradeNote: The threadpool_* options in stasis.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 stasis. DeveloperNote: The taskpool API has been added for common usage of a pool of taskprocessors. It is suggested to use this API instead of the threadpool+taskprocessor approach.	11 months ago

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34263 Commits (487bf8af261b475141cb0ba83aed9fc5b0033868) All Branches Search

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