34241 Commits (21df9edc380fcf17ffff2fa5cacda601fe5c19b6)

 

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Author	SHA1	Message	Date
George Joseph	21df9edc38	taskpool: Fix some references to threadpool that should be taskpool. ... Resolves: #1478	3 months ago
Anthony Minessale	fab46493b3	Update contact information for anthm	3 months ago
George Joseph	a63eec2fbb	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	3 months ago
Naveen Albert	6a82f4c390	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	3 months ago
Naveen Albert	2041a4645f	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.	3 months ago
Naveen Albert	88680c3e7e	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	3 months ago
Igor Goncharovsky	5177662990	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.	3 months ago
Naveen Albert	cf9c107e4d	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	3 months ago
Naveen Albert	9a2a853fee	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.	3 months ago
Naveen Albert	341cb16378	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.	3 months ago
Naveen Albert	b8c9440e19	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	3 months ago
Naveen Albert	de46164a73	res_tonedetect: Fix formatting of XML documentation. ... Fix the indentation in the documentation for the variable list. Resolves: #1507	3 months ago
Naveen Albert	3c21fbf69c	res_fax: Add XML documentation for channel variables. ... Document the channel variables currently set by SendFAX and ReceiveFAX. Resolves: #1505	3 months ago
George Joseph	c597c8b5c0	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	3 months ago
Naveen Albert	a6b15bfb68	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.	3 months ago
Naveen Albert	e57f10bbe5	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	3 months ago
Naveen Albert	037bf58c78	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	3 months ago
Naveen Albert	45646be97a	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	a96d7fcfaf	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	559ea45ddd	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	2c97fd3ea4	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	ece6ed9459	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	ff80666aac	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
Allan Nathanson	7bd30279de	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	4 months ago
George Joseph	32c41efa04	res_rtp_asterisk.c: Use rtp->dtls in __rtp_sendto when rtcp mux is used. ... In __rtp_sendto(), the check for DTLS negotiation completion for rtcp packets needs to use the rtp->dtls structure instead of rtp->rtcp->dtls when AST_RTP_INSTANCE_RTCP_MUX is set. Resolves: #1474	4 months ago
Bastian Triller	a0f1e460fe	Fix some doxygen, typos and whitespace	4 months ago
Sven Kube	9e50fb3880	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
George Joseph	896850ca35	chan_websocket: Fix codec validation and add passthrough option. ... * Fixed an issue in webchan_write() where we weren't detecting equivalent codecs properly. * Added the "p" dialstring option that puts the channel driver in "passthrough" mode where it will not attempt to re-frame or re-time media coming in over the websocket from the remote app. This can be used for any codec but MUST be used for codecs that use packet headers or whose data stream can't be broken up on arbitrary byte boundaries. In this case, the remote app is fully responsible for correctly framing and timing media sent to Asterisk and the MEDIA text commands that could be sent over the websocket are disabled. Currently, passthrough mode is automatically set for the opus, speex and g729 codecs. * Now calling ast_set_read_format() after ast_channel_set_rawreadformat() to ensure proper translation paths are set up when switching between native frames and slin silence frames. This fixes an issue with codec errors when transcode_via_sln=yes. Resolves: #1462	4 months ago
phoneben	af67321134	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
Sean Bright	7b96fff129	app_externalivr: Prevent out-of-bounds read during argument processing. ... Resolves: #1422	4 months ago
Naveen Albert	3a7d622c63	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.	4 months ago
Joshua C. Colp	77d630f57a	taskpool: Update versions for taskpool stasis options.	4 months ago
Joshua C. Colp	e9ee9d7d98	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.	4 months ago
George Joseph	390d8f98c2	res_ari: Ensure outbound websocket config has a websocket_client_id. ... Added a check to outbound_websocket_apply() that makes sure an outbound websocket config object in ari.conf has a websocket_client_id parameter. Resolves: #1457	4 months ago
Naveen Albert	4aad564b93	app_adsiprog: Fix possible NULL dereference. ... get_token can return NULL, but process_token uses this result without checking for NULL; as elsewhere, check for a NULL result to avoid possible NULL dereference. Resolves: #1419	4 months ago
Nathan Monfils	170048a41e	manager.c: Fix presencestate object leak ... ast_presence_state allocates subtype and message. We straightforwardly need to clean those up.	4 months ago
Sean Bright	1b42eda449	audiohook.c: Ensure correct AO2 reference is dereffed. ... Part of #1440.	4 months ago
Naveen Albert	7ffde57474	res_cliexec: Remove unnecessary casts to char*. ... Resolves: #1436	4 months ago
Ben Ford	8add74c462	rtp_engine.c: Add exception for comfort noise payload. ... In a previous commit, a change was made to ast_rtp_codecs_payload_code_tx_sample_rate to check for differing sample rates. This ended up returning an invalid payload int for comfort noise. A check has been added that returns early if the payload is in fact supposed to be comfort noise. Fixes: #1340	4 months ago
Naveen Albert	971beef41c	pbx_variables.c: Create real channel for "dialplan eval function". ... "dialplan eval function" has been using a dummy channel for function evaluation, much like many of the unit tests. However, sometimes, this can cause issues for functions that are not expecting dummy channels. As an example, ast_channel_tech(chan) is NULL on such channels, and ast_channel_tech(chan)->type consequently results in a NULL dereference. Normally, functions do not worry about this since channels executing dialplan aren't dummy channels. While some functions are better about checking for these sorts of edge cases, use a real channel with a dummy technology to make this CLI command inherently safe for any dialplan function that could be evaluated from the CLI. Resolves: #1434	4 months ago
Joe Garlick	0deac782db	chan_websocket.c: Add DTMF messages ... Added DTMF messages to the chan_websocket feature. When a user presses DTMF during a call over chan_websocket it will send a message like: "DTMF_END digit:1" Resolves: https://github.com/asterisk/asterisk-feature-requests/issues/70	4 months ago
Igor Goncharovsky	f77e76bab1	app_queue.c: Add new global 'log_unpause_on_reason_change' ... In many asterisk-based systems, the pause reason is used to separate pauses by type,and logically, changing the reason defines two intervals that should be accounted for separately. The introduction of a new option allows me to separate the intervals of operator inactivity in the log by the event of unpausing. UserNote: Add new global option 'log_unpause_on_reason_change' that is default disabled. When enabled cause addition of UNPAUSE event on every re-PAUSE with reason changed.	4 months ago
Igor Goncharovsky	39c677e0df	app_waitforsilence.c: Use milliseconds to calculate timeout time ... The functions WaitForNoise() and WaitForSilence() use the time() functions to calculate elapsed time, which causes the timer to fire on a whole second boundary, and the actual function execution time to fire the timer may be 1 second less than expected. This fix replaces time() with ast_tvnow(). Fixes: #1401	4 months ago
Artem Umerov	96e8535e84	Fix missing ast_test_flag64 in extconf.c ... Fix missing ast_test_flag64 after 43bf8a4ded	4 months ago
Naveen Albert	6fb811a590	pbx_builtins: Allow custom tone for WaitExten. ... Currently, the 'd' option will play dial tone while waiting for digits. Allow it to accept an argument for any tone from indications.conf. Resolves: #1396 UserNote: The tone used while waiting for digits in WaitExten can now be overridden by specifying an argument for the 'd' option.	4 months ago
Naveen Albert	0a46be95cb	res_tonedetect: Add option for TONE_DETECT detection to auto stop. ... One of the problems with TONE_DETECT as it was originally written is that if a tone is detected multiple times, it can trigger the redirect logic multiple times as well. For example, if we do an async goto in the dialplan after detecting a tone, because the detector is still active until explicitly disabled, if we detect the tone again, we will branch again and start executing that dialplan a second time. This is rarely ever desired behavior, and can happen if the detector is not removed quickly enough. Add a new option, 'e', which automatically disables the detector once the desired number of matches have been heard. This eliminates the potential race condition where previously the detector would need to be disabled immediately, but doing so quickly enough was not guaranteed. This also allows match criteria to be retained longer if needed, so the detector does not need to be destroyed prematurely. Resolves: #1390 UserNote: The 'e' option for TONE_DETECT now allows detection to be disabled automatically once the desired number of matches have been fulfilled, which can help prevent race conditions in the dialplan, since TONE_DETECT does not need to be disabled after a hit.	4 months ago
Stuart Henderson	11e3567200	app_queue: fix comparison for announce-position-only-up ... Numerically comparing that the current queue position is less than last_pos_said can only be done after at least one announcement has been made, otherwise last_pos_said is at the default (0). Fixes: #1386	4 months ago
George Joseph	02993717b0	res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL. ... In the highly-unlikely event that get_authorization_hdr() couldn't find an Authorization header in a request, trying to get the digest algorithm would cauase a SEGV. We now check that we have an auth header that matches the realm before trying to get the algorithm from it. Resolves: #GHSA-64qc-9x89-rx5j	5 months ago
Alexei Gradinari	336f6bd627	sorcery: Prevent duplicate objects and ensure missing objects are created on update ... This patch resolves two issues in Sorcery objectset handling with multiple backends: 1. Prevent duplicate objects: When an object exists in more than one backend (e.g., a contact in both 'astdb' and 'realtime'), the objectset previously returned multiple instances of the same logical object. This caused logic failures in components like the PJSIP registrar, where duplicate contact entries led to overcounting and incorrect deletions, when max_contacts=1 and remove_existing=yes. This patch ensures only one instance of an object with a given key is added to the objectset, avoiding these duplicate-related side effects. 2. Ensure missing objects are created: When using multiple writable backends, a temporary backend failure can lead to objects missing permanently from that backend. Currently, .update() silently fails if the object is not present, and no .create() is attempted. This results in inconsistent state across backends (e.g. astdb vs. realtime). This patch introduces a new global option in sorcery.conf: [general] update_or_create_on_update_miss = yes|no Default: no (preserves existing behavior). When enabled: if .update() fails with no data found, .create() is attempted in that backend. This ensures that objects missing due to temporary backend outages are re-synchronized once the backend is available again. Added a new CLI command: sorcery show settings Displays global Sorcery settings, including the current value of update_or_create_on_update_miss. Updated tests to validate both flag enabled/disabled behavior. Fixes: #1289 UserNote: Users relying on Sorcery multiple writable backends configurations (e.g., astdb + realtime) may now enable update_or_create_on_update_miss = yes in sorcery.conf to ensure missing objects are recreated after temporary backend failures. Default behavior remains unchanged unless explicitly enabled.	5 months ago
Naveen Albert	c77247001b	sig_analog: Skip Caller ID spill if usecallerid=no. ... If Caller ID is disabled for an FXS port, then we should not send any Caller ID spill on the line, as we have no Caller ID information that we can/should be sending. Resolves: #1394	5 months ago