4459 Commits (cc5542e753aeb85c420698ab6797066ea8da8148)

Author	SHA1	Message	Date
George Joseph	790f668057	chan_pjsip: Set correct cause codes for non-2XX responses. ... Redirects initiated by 302 response codes weren't handled correctly when setting the hangup cause code and tech cause code on the responding channel. They're now set to 23 (REDIRECTED_TO_NEW_DESTINATION) and 302 (Moved permanently). Other non-2XX response codes also had issues. A new API ast_channel_dialed_causes_iterator() was added to retrieve the hangup cause codes for a channel. chan_pjsip_session_end() in chan_pjsip has been refactored to set the correct cause codes on a channel based on the cause codes added by chan_pjsip_incoming_response_update_cause(). Copious amounts of debugging and comments were also added. Resolves: #1819	5 months ago
Naveen Albert	14ddbf022a	dsp.c: Add support for detecting R2 signaling tones. ... Extend the existing DTMF/MF tone detection support by adding support for R2 tones, another variant of MF (R1) signaling. Both forward and backward signaling are supported. Resolves: #1804	5 months ago
George Joseph	d0a30ae399	docs: Add "Provided-by" to doc XML and CLI output. ... For application, function, manager, managerEvent, managerEventInstance and info XML documentation nodes, the make_xml_documentation script will add a "module" attribute if not already present. For XML in separate "*_doc.xml" files, the script figures out the correct module name. For documentation in the "main" directory, the module name is set to "builtin". The CLI handlers for "core show application", "core show function", "manager show command" and "manager show event", have been updated to show the following after the Synopsis... ``` [Provided By] <modulename> ``` For modules that provide additional "info" elements (like the technologies do for Dial), the providing module has also been added. ``` Technology: WebSocket Provided by: chan_websocket WebSocket Dial Strings: ... ``` UserNote: The CLI help for applications, functions, manager commands and manager events now shows the module that provides its functionality.	5 months ago
George Joseph	98ac114678	CDR/CEL Custom Performance Improvements ... There is a LOT of work in this commit but the TL;DR is that it takes CEL processing from using 38% of the CPU instructions used by a call, which is more than that used by the call processing itself, down to less than 10% of the instructions. So here's the deal... cdr_custom, cdr_sqlite3_custom, cel_custom and cel_sqlite3_custom all shared one ugly trait...they all used ast_str_substitute_variables() or pbx_substitute_variables_helper() to resolve the dialplan functions used in their config files. Not only are they both extraordinarily expensive, they both require a dummy channel to be allocated and destroyed for each record written. For CDRs, that's not too bad because we only write one CDR per call. For CELs however, it's a disaster. As far as source code goes, the modules basically all did the same thing. Unfortunately, they did it badly. The config files simply contained long opaque strings which were intepreted by ast_str_substitute_variables() or pbx_substitute_variables_helper(), the very functions that ate all the instructions. This meant introducing a new "advanced" config format much like the advanced manager event filtering added to manager.conf in 2024. Fortunately however, if the legacy config was recognizable, we were able to parse it as an advanced config and gain the benefit. If not, then it goes the legacy, and very expensive, route. Given the commonality among the modules, instead of making the same improvements to 4 modules then trying to maintain them over time, a single module "res_cdrel_custom" was created that contains all of the common code. A few bonuses became possible in the process... * The cdr_custom and cel_custom modules now support JSON formatted output. * The cdr_sqlite_custom and cel_sqlite3_custom modules no longer have to share an Sqlite3 database. Summary of changes: A new module "res/res_cdrel_custom.c" has been created and the existing cdr_custom, cdr_sqlite3_custom, cel_custom and cel_sqlite3_custom modules are now just stubs that call the code in res_cdrel_custom. res_cdrel_custom contains: * A common configuration facility. * Getters for both CDR and CEL fields that share the same abstraction. * Formatters for all data types found in the ast_cdr and ast_event structures that share the same abstraction. * Common writers for the text file and database backends that, you guessed it, share the same abstraction. The result is that while there is certainly a net increase in the number of lines in the code base, most of it is in the configuration handling at load-time. The run-time instruction path length is now significanty shorter. ``` Scenario Instructions Latency ===================================================== CEL pre changes 38.49% 37.51% CEL Advanced 9.68% 6.06% CEL Legacy (auto-conv to adv) 9.95% 6.13% CEL Sqlite3 pre changes 39.41% 39.90% CEL Sqlite3 Advanced 25.68% 24.24% CEL Sqlite3 Legacy (auto conv) 25.88% 24.53% CDR pre changes 4.79% 2.95% CDR Advanced 0.79% 0.47% CDR Legacy (auto conv to adv) 0.86% 0.51% CDR Sqlite3 pre changes 4.47% 2.89% CEL Sqlite3 Advanced 2.16% 1.29% CEL Sqlite3 Legacy (auto conv) 2.19% 1.30% ``` Notes: * We only write one CDR per call but every little bit helps. * Sqlite3 still takes a fair amount of resources but the new config makes a decent improvement. * Legacy configs that we can't auto convert will still take the "pre changes" path. If you're interested in more implementation details, see the comments at the top of the res_cdrel_custom.c file. One minor fix to CEL is also included...Although TenantID was added to the ast_event structure, it was always rendered as an empty string. It's now properly rendered. UserNote: Significant performance improvements have been made to the cdr_custom, cdr_sqlite3_custom, cel_custom and cel_sqlite3_custom modules. See the new sample config files for those modules to see how to benefit from them.	5 months ago
Prashant Srivastav	5938c820a1	fix: Add macOS (Darwin) compatibility for building Asterisk ... - Makefile: Skip /usr/lib/bundle1.o on modern macOS (doesn't exist) - Makefile.rules: Skip -fno-partial-inlining for clang (gcc-only flag) - include/asterisk/utils.h: Use asterisk/endian.h instead of <endian.h> - main/Makefile: Add Darwin-specific pjproject linking with -force_load - main/strcompat.c: Include poll-compat.h, use ast_poll() - main/xml.c: Add ASTMM_LIBC ASTMM_IGNORE for libxml2 compatibility - res/res_pjsip/config_transport.c: Define TCP keepalive constants for macOS Tested on macOS Darwin 25.2.0 (Apple Silicon ARM64)	5 months ago
Joshua C. Colp	ab04bf3b39	endpoints: Allow access to latest snapshot directly. ... This change adds an API call to allow direct access to the latest snapshot of an ast_endpoint. This is then used by chan_pjsip when calculating device state, eliminating the need to access the cache which would incur a container find and access.	5 months ago
Naveen Albert	c3db651b23	app_dial, utils.h: Avoid old style declaration and discarded qualifier. ... * app_dial: Use const char* for fixed strings. * utils.h: inline should come before return type. Resolves: #1755	5 months ago
Sean Bright	0e6bdea103	cli.c: Allow 'channel request hangup' to accept patterns. ... This extends 'channel request hangup' to accept multiple channel names, a POSIX Extended Regular Expression, a glob-like pattern, or a combination of all of them. UserNote: The 'channel request hangup' CLI command now accepts multiple channel names, POSIX Extended Regular Expressions, glob-like patterns, or a combination of all of them. See the CLI command 'core show help channel request hangup' for full details.	8 months ago
George Joseph	10060d8dde	chan_websocket: Use the channel's ability to poll fds for the websocket read. ... We now add the websocket's file descriptor to the channel's fd array and let it poll for data availability instead if having a dedicated thread that does the polling. This eliminates the thread and allows removal of most explicit locking since the core channel code will lock the channel to prevent simultaneous calls to webchan_read, webchan_hangup, etc. While we were here, the hangup code was refactored to use ast_hangup_with_cause instead of directly queueing an AST_CONTROL_HANGUP frame. This allows us to set hangup causes and generate snapshots. For a bit of extra debugging, a table of websocket close codes was added to http_websocket.h with an accompanying "to string" function added to res_http_websocket.c Resolves: #1683	8 months ago
George Joseph	8ca3a4978d	res_geolocation: Fix multiple issues with XML generation. ... * 3d positions were being rendered without an enclosing `<gml:pos>` element resulting in invalid XML. * There was no way to set the `id` attribute on the enclosing `tuple`, `device` and `person` elements. * There was no way to set the value of the `deviceID` element. * Parsing of degree and radian UOMs was broken resulting in them appearing outside an XML element. * The UOM schemas for degrees and radians were reversed. * The Ellipsoid shape was missing and the Ellipse shape was defined multiple times. * The `crs` location_info parameter, although documented, didn't work. * The `pos3d` location_info parameter appears in some documentation but wasn't being parsed correctly. * The retransmission-allowed and retention-expiry sub-elements of usage-rules were using the `gp` namespace instead of the `gbp` namespace. In addition to fixing the above, several other code refactorings were performed and the unit test enhanced to include a round trip XML -> eprofile -> XML validation. Resolves: #1667 UserNote: Geolocation: Two new optional profile parameters have been added. * `pidf_element_id` which sets the value of the `id` attribute on the top-level PIDF-LO `device`, `person` or `tuple` elements. * `device_id` which sets the content of the `<deviceID>` element. Both parameters can include channel variables. UpgradeNote: Geolocation: In order to correct bugs in both code and documentation, the following changes to the parameters for GML geolocation locations are now in effect: * The documented but unimplemented `crs` (coordinate reference system) element has been added to the location_info parameter that indicates whether the `2d` or `3d` reference system is to be used. If the crs isn't valid for the shape specified, an error will be generated. The default depends on the shape specified. * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs. If crs isn't specified, it will default to `2d` for these shapes. The Sphere, Ellipsoid and Prism shapes MUST use a `3d` crs. If crs isn't specified, it will default to `3d` for these shapes. The Point and Polygon shapes may use either crs. The default crs is `2d` however so if `3d` positions are used, the crs must be explicitly set to `3d`. * The `geoloc show gml_shape_defs` CLI command has been updated to show which coordinate reference systems are valid for each shape. * The `pos3d` element has been removed in favor of allowing the `pos` element to include altitude if the crs is `3d`. The number of values in the `pos` element MUST be 2 if the crs is `2d` and 3 if the crs is `3d`. An error will be generated for any other combination. * The angle unit-of-measure for shapes that use angles should now be included in the respective parameter. The default is `degrees`. There were some inconsistent references to `orientation_uom` in some documentation but that parameter never worked and is now removed. See examples below. Examples... ``` location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20" location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620" location_info = shape="Point", pos="39.0 -105.0" location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20" semiMinorAxis="10", verticalAxis="0", orientation="25 degrees" pidf_element_id = ${CHANNEL(name)}-${EXTEN} device_id = mac:001122334455 Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN}) ```	8 months ago
Maximilian Fridrich	3fa6e90d2d	res_pjsip: Introduce redirect module for handling 3xx responses ... This commit introduces a new redirect handling module that provides infrastructure for following SIP 3xx redirect responses. The redirect functionality respects the endpoint's redirect_method setting and only follows redirects when set to 'uri_pjsip'. This infrastructure can be used by any PJSIP module that needs to handle 3xx redirect responses.	8 months ago
Justin T. Gibbs	bd8a510a6b	rtp/rtcp: Configure dual-stack behavior via IPV6_V6ONLY ... Dual-stack behavior (simultaneous listening for IPV4 and IPV6 connections on a single socket) is required by Asterisk's ICE implementation. On systems with the IPV6_V6ONLY sockopt, set the option to 0 (dual-stack enabled) when binding to the IPV6 any address. This ensures correct behavior regardless of the system's default dual-stack configuration.	8 months ago
Joshua C. Colp	44a68fb442	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.	8 months ago
Sean Bright	45ffbb3f3a	Revert "func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()" ... This reverts commit 5177662990. For rationale, see #1621 and #1606	8 months ago
Mike Bradeen	b42bf1d319	taskprocessors: Improve logging and add new cli options ... This change makes some small changes to improve log readability in addition to the following changes: Modified 'core show taskprocessors' to now show Low time and High time for task execution. New command 'core show taskprocessor name <taskprocessor-name>' to dump taskprocessor info and current queue. Addionally, a new test was added to demonstrate the 'show taskprocessor name' functionality: test execute category /main/taskprocessor/ name taskprocessor_cli_show Setting 'core set debug 3 taskprocessor.c' will now log pushed tasks. (Warning this is will cause extremely high levels of logging at even low traffic levels.) Resolves: #1566 UserNote: New CLI command has been added - core show taskprocessor name <taskprocessor-name>	8 months ago
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
Joshua C. Colp	54da3e1b9e	Revert "pjsip: Move from threadpool to taskpool" ... This reverts commit bb6b76c2d8.	10 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.	10 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	10 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.	10 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.	10 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.	10 months ago
George Joseph	b3e5ca59a0	taskpool: Fix some references to threadpool that should be taskpool. ... Resolves: #1478	10 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.	10 months ago
Bastian Triller	bb2b26e8ca	Fix some doxygen, typos and whitespace	10 months ago
Joshua C. Colp	f3f005e49e	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.	10 months ago
Alexei Gradinari	44034cf841	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. (cherry picked from commit 658b775fd6)	12 months ago
George Joseph	7cc840ef77	chan_websocket: Allow additional URI parameters to be added to the outgoing URI. ... * Added a new option to the WebSocket dial string to capture the additional URI parameters. * Added a new API ast_uri_verify_encoded() that verifies that a string either doesn't need URI encoding or that it has already been encoded. * Added a new API ast_websocket_client_add_uri_params() to add the params to the client websocket session. * Added XML documentation that will show up with `core show application Dial` that shows how to use it. Resolves: #1352 UserNote: A new WebSocket channel driver option `v` has been added to the Dial application that allows you to specify additional URI parameters on outgoing connections. Run `core show application Dial` from the Asterisk CLI to see how to use it. (cherry picked from commit 15371efeab)	12 months ago
Sven Kube	c4103f259a	ARI: Add command to indicate progress to a channel ... Adds an ARI command to send a progress indication to a channel. DeveloperNote: A new ARI endpoint is available at `/channels/{channelId}/progress` to indicate progress to a channel. (cherry picked from commit f9ab56b1d8)	12 months ago
Sperl Viktor	fcb35f97b9	cel: Add STREAM_BEGIN, STREAM_END and DTMF event types. ... Fixes: #1280 UserNote: Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types in cel.conf will log media files and music on hold played to each channel. The STREAM_BEGIN event's extra field will contain a JSON with the file details (path, format and language), or the class name, in case of music on hold is played. The DTMF event's extra field will contain a JSON with the digit and the duration in milliseconds. (cherry picked from commit f516f07fd2)	12 months ago
zhou_jiajian	3cedbab993	cdr: add CANCEL dispostion in CDR ... In the original implementation, both CANCEL and NO ANSWER states were consolidated under the NO ANSWER disposition. This patch introduces a separate CANCEL disposition, with an optional configuration switch to enable this new disposition. Resolves: #1323 UserNote: A new CDR option "canceldispositionenabled" has been added that when set to true, the NO ANSWER disposition will be split into two dispositions: CANCEL and NO ANSWER. The default value is 'no' (cherry picked from commit 01661398a3)	12 months ago
George Joseph	9eef99f8f3	options: Change ast_options from ast_flags to ast_flags64. ... DeveloperNote: The 32-bit ast_options has no room left to accomodate new options and so has been converted to an ast_flags64 structure. All internal references to ast_options have been updated to use the 64-bit flag manipulation macros. External module references to the 32-bit ast_options should continue to work on little-endian systems because the least-significant bytes of a 64 bit integer will be in the same location as a 32-bit integer. Because that's not the case on big-endian systems, we've swapped the bytes in the flags manupulation macros on big-endian systems so external modules should still work however you are encouraged to test. (cherry picked from commit 709be42bc3)	12 months ago
Alexei Gradinari	5bdc661745	res_config_odbc: Prevent Realtime fallback on record-not-found (SQL_NO_DATA) ... This patch fixes an issue in the ODBC Realtime engine where Asterisk incorrectly falls back to the next configured backend when the current one returns SQL_NO_DATA (i.e., no record found). This is a logical error and performance risk in multi-backend configurations. Solution: Introduced CONFIG_RT_NOT_FOUND ((void *)-1) as a special return marker. ODBC Realtime backend now return CONFIG_RT_NOT_FOUND when no data is found. Core engine stops iterating on this marker, avoiding unnecessary fallback. Notes: Other Realtime backends (PostgreSQL, LDAP, etc.) can be updated similarly. This patch only covers ODBC. Fixes: #1305 (cherry picked from commit dfe25fbc8a)	12 months ago
George Joseph	d40d5c999a	Media over Websocket Channel Driver ... * Created chan_websocket which can exchange media over both inbound and outbound websockets which the driver will frame and time. See http://s.asterisk.net/mow for more information. * res_http_websocket: Made defines for max message size public and converted a few nuisance verbose messages to debugs. * main/channel.c: Changed an obsolete nuisance error to a debug. * ARI channels: Updated externalMedia to include chan_websocket as a supported transport. UserNote: A new channel driver "chan_websocket" is now available. It can exchange media over both inbound and outbound websockets and will both frame and re-time the media it receives. See http://s.asterisk.net/mow for more information. UserNote: The ARI channels/externalMedia API now includes support for the WebSocket transport provided by chan_websocket. (cherry picked from commit 733196abf9)	12 months ago
mkmer	798fd4eb63	utils.h: Add rounding to float conversion to int. ... Quote from an audio engineer NR9V: There is a minor issue of a small amount of crossover distortion though as a result of `ast_slinear_saturated_multiply_float()` not rounding the float. This could result in some quiet but potentially audible distortion artifacts in lower volume parts of the signal. If you have for example a sign wave function with a max amplitude of just a few samples, all samples between -1 and 1 will be truncated to zero, resulting in the waveform no longer being a sine wave and in harmonic distortion. Resolves: #1176 (cherry picked from commit 6d06d1765f)	12 months ago
Tinet-mucw	9803156223	pbx.c: when set flag AST_SOFTHANGUP_ASYNCGOTO, ast_explicit_goto should return -1. ... Under certain circumstances the context/extens/prio are set in the ast_async_goto, for example action Redirect. In the situation that action Redirect is broken by GotoIf this info is changed. that will causes confusion in dialplan execution. Resolves: #1273 (cherry picked from commit bf9a6d80f8)	12 months ago
George Joseph	0b29f5c60c	ARI Outbound Websockets ... Asterisk can now establish websocket sessions _to_ your ARI applications as well as accepting websocket sessions _from_ them. Full details: http://s.asterisk.net/ari-outbound-ws Code change summary: * Added an ast_vector_string_join() function, * Added ApplicationRegistered and ApplicationUnregistered ARI events. * Converted res/ari/config.c to use sorcery to process ari.conf. * Added the "outbound-websocket" ARI config object. * Refactored res/ari/ari_websockets.c to handle outbound websockets. * Refactored res/ari/cli.c for the sorcery changeover. * Updated res/res_stasis.c for the sorcery changeover. * Updated apps/app_stasis.c to allow initiating per-call outbound websockets. * Added CLI commands to manage ARI websockets. * Added the new "outbound-websocket" object to ari.conf.sample. * Moved the ARI XML documentation out of res_ari.c into res/ari/ari_doc.xml UserNote: Asterisk can now establish websocket sessions _to_ your ARI applications as well as accepting websocket sessions _from_ them. Full details: http://s.asterisk.net/ari-outbound-ws	12 months ago
George Joseph	b08e093072	res_websocket_client: Create common utilities for websocket clients. ... Since multiple Asterisk capabilities now need to create websocket clients it makes sense to create a common set of utilities rather than making each of those capabilities implement their own. * A new configuration file "websocket_client.conf" is used to store common client parameters in named configuration sections. * APIs are provided to list and retrieve ast_websocket_client objects created from the named configurations. * An API is provided that accepts an ast_websocket_client object, connects to the remote server with retries and returns an ast_websocket object. TLS is supported as is basic authentication. * An observer can be registered to receive notification of loaded or reloaded client objects. * An API is provided to compare an existing client object to one just reloaded and return the fields that were changed. The caller can then decide what action to take based on which fields changed. Also as part of thie commit, several sorcery convenience macros were created to make registering common object fields easier. UserNote: A new module "res_websocket_client" and config file "websocket_client.conf" have been added to support several upcoming new capabilities that need common websocket client configuration.	12 months ago
George Joseph	2939725743	asterisk.c: Add option to restrict shell access from remote consoles. ... UserNote: A new asterisk.conf option 'disable_remote_console_shell' has been added that, when set, will prevent remote consoles from executing shell commands using the '!' prefix. Resolves: #GHSA-c7p6-7mvq-8jq2	12 months ago
George Joseph	6f2b99bfa6	Alternate Channel Storage Backends ... Full details: http://s.asterisk.net/dc679ec3 The previous proof-of-concept showed that the cpp_map_name_id alternate storage backed performed better than all the others so this final PR adds only that option. You still need to enable it in menuselect under the "Alternate Channel Storage Backends" category. To select which one is used at runtime, set the "channel_storage_backend" option in asterisk.conf to one of the values described in asterisk.conf.sample. The default remains "ao2_legacy". UpgradeNote: With this release, you can now select an alternate channel storage backend based on C++ Maps. Using the new backend may increase performance and reduce the chances of deadlocks on heavily loaded systems. For more information, see http://s.asterisk.net/dc679ec3	12 months ago
Naveen Albert	6002b6fdcb	sig_analog: Add Call Waiting Deluxe support. ... Adds support for Call Waiting Deluxe options to enhance the current call waiting feature. As part of this change, a mechanism is also added that allows a channel driver to queue an audio file for Dial() to play, which is necessary for the announcement function. ASTERISK-30373 #close Resolves: #271 UserNote: Call Waiting Deluxe can now be enabled for FXS channels by enabling its corresponding option.	12 months ago
George Joseph	7e8f566b49	lock.h: Add include for string.h when DEBUG_THREADS is defined. ... When DEBUG_THREADS is defined, lock.h uses strerror(), which is defined in the libc string.h file, to print warning messages. If the including source file doesn't include string.h then strerror() won't be found and and compile errors will be thrown. Since lock.h depends on this, string.h is now included from there if DEBUG_THREADS is defined. This way, including source files don't have to worry about it.	12 months ago
Albrecht Oster	d7b8f1a3a1	res_pjproject: Fix DTLS client check failing on some platforms ... Certain platforms (mainly BSD derivatives) have an additional length field in `sockaddr_in6` and `sockaddr_in`. `ast_sockaddr_from_pj_sockaddr()` does not take this field into account when copying over values from the `pj_sockaddr` into the `ast_sockaddr`. The resulting `ast_sockaddr` will have an uninitialized value for `sin6_len`/`sin_len` while the other `ast_sockaddr` (not converted from a `pj_sockaddr`) to check against in `ast_sockaddr_pj_sockaddr_cmp()` has the correct length value set. This has the effect that `ast_sockaddr_cmp()` will always indicate an address mismatch, because it does a bitwise comparison, and all DTLS packets are dropped even if addresses and ports match. `ast_sockaddr_from_pj_sockaddr()` now checks whether the length fields are available on the current platform and sets the values accordingly. Resolves: #505 (cherry picked from commit 315289b87a)	1 year ago
George Joseph	b4378ff25e	Prequisites for ARI Outbound Websockets ... stasis: * Added stasis_app_is_registered(). * Added stasis_app_control_mark_failed(). * Added stasis_app_control_is_failed(). * Fixed res_stasis_device_state so unsubscribe all works properly. * Modified stasis_app_unregister() to unsubscribe from all event sources. * Modified stasis_app_exec to return -1 if stasis_app_control_is_failed() returns true. http: * Added ast_http_create_basic_auth_header(). md5: * Added define for MD5_DIGEST_LENGTH. tcptls: * Added flag to ast_tcptls_session_args to suppress connection log messages to give callers more control over logging. http_websocket: * Add flag to ast_websocket_client_options to suppress connection log messages to give callers more control over logging. * Added username and password to ast_websocket_client_options to support outbound basic authentication. * Added ast_websocket_result_to_str(). (cherry picked from commit 9d75b0447f)	1 year ago
George Joseph	2a55f064f3	ARI: REST over Websocket ... This commit adds the ability to make ARI REST requests over the same websocket used to receive events. For full details on how to use the new capability, visit... https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/ Changes: * Added utilities to http.c: * ast_get_http_method_from_string(). * ast_http_parse_post_form(). * Added utilities to json.c: * ast_json_nvp_array_to_ast_variables(). * ast_variables_to_json_nvp_array(). * Added definitions for new events to carry REST responses. * Created res/ari/ari_websocket_requests.c to house the new request handlers. * Moved non-event specific code out of res/ari/resource_events.c into res/ari/ari_websockets.c * Refactored res/res_ari.c to move non-http code out of ast_ari_callback() (which is http specific) and into ast_ari_invoke() so it can be shared between both the http and websocket transports. UpgradeNote: This commit adds the ability to make ARI REST requests over the same websocket used to receive events. See https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/ (cherry picked from commit 1442c17141)	1 year ago
mkmer	68d9a69be3	audiohook.c: Add ability to adjust volume with float ... Add the capability to audiohook for float type volume adjustments. This allows for adjustments to volume smaller than 6dB. With INT adjustments, the first step is 2 which converts to ~6dB (or 1/2 volume / double volume depending on adjustment sign). 3dB is a typical adjustment level which can now be accommodated with an adjustment value of 1.41. This is accomplished by the following: Convert internal variables to type float. Always use ast_frame_adjust_volume_float() for adjustments. Cast int to float in original functions ast_audiohook_volume_set(), and ast_volume_adjust(). Cast float to int in ast_audiohook_volume_get() Add functions ast_audiohook_volume_get_float, ast_audiohook_volume_set_float, and ast_audiohook_volume_adjust_float. This update maintains 100% backward compatibility. Resolves: #1171 (cherry picked from commit ecd1a727e8)	1 year ago
Florent CHAUVEAU	a15e4043ce	audiosocket: added support for DTMF frames ... Updated the AudioSocket protocol to allow sending DTMF frames. AST_FRAME_DTMF frames are now forwarded to the server, in addition to AST_FRAME_AUDIO frames. A new payload type AST_AUDIOSOCKET_KIND_DTMF with value 0x03 was added to the protocol. The payload is a 1-byte ascii representing the DTMF digit (0-9,*,#...). UserNote: The AudioSocket protocol now forwards DTMF frames with payload type 0x03. The payload is a 1-byte ascii representing the DTMF digit (0-9,*,#...). (cherry picked from commit c2444b188a)	1 year ago
Norm Harrison	352a72c9b5	asterisk/channel.h: fix documentation for 'ast_waitfor_nandfds()' ... Co-authored-by: Florent CHAUVEAU <florentch@pm.me> (cherry picked from commit 6c6195ce4d)	1 year ago
Norm Harrison	d045f290b7	audiosocket: fix timeout, fix dialplan app exit, server address in logs ... - Correct wait timeout logic in the dialplan application. - Include server address in log messages for better traceability. - Allow dialplan app to exit gracefully on hangup messages and socket closure. - Optimize I/O by reducing redundant read()/write() operations. Co-authored-by: Florent CHAUVEAU <florentch@pm.me> (cherry picked from commit a6ef41e4ca)	1 year ago
George Joseph	de3d6c03e3	bridging: Fix multiple bridging issues causing SEGVs and FRACKs. ... Issues: * The bridging core allowed multiple bridges to be created with the same unique bridgeId at the same time. Only the last bridge created with the duplicate name was actually saved to the core bridges container. * The bridging core was creating a stasis topic for the bridge and saving it in the bridge->topic field but not increasing its reference count. In the case where two bridges were created with the same uniqueid (which is also the topic name), the second bridge would get the _existing_ topic the first bridge created. When the first bridge was destroyed, it would take the topic with it so when the second bridge attempted to publish a message to it it either FRACKed or SEGVd. * The bridge destructor, which also destroys the bridge topic, is run from the bridge manager thread not the caller's thread. This makes it possible for an ARI developer to create a new one with the same uniqueid believing the old one was destroyed when, in fact, the old one's destructor hadn't completed. This could cause the new bridge to get the old one's topic just before the topic was destroyed. When the new bridge attempted to publish a message on that topic, asterisk could either FRACK or SEGV. * The ARI bridges resource also allowed multiple bridges to be created with the same uniqueid but it kept the duplicate bridges in its app_bridges container. This created a situation where if you added two bridges with the same "bridge1" uniqueid, all operations on "bridge1" were performed on the first bridge created and the second was basically orphaned. If you attempted to delete what you thought was the second bridge, you actually deleted the first one created. Changes: * A new API `ast_bridge_topic_exists(uniqueid)` was created to determine if a topic already exists for a bridge. * `bridge_base_init()` in bridge.c and `ast_ari_bridges_create()` in resource_bridges.c now call `ast_bridge_topic_exists(uniqueid)` to check if a bridge with the requested uniqueid already exists and will fail if it does. * `bridge_register()` in bridges.c now checks the core bridges container to make sure a bridge doesn't already exist with the requested uniqueid. Although most callers of `bridge_register()` will have already called `bridge_base_init()`, which will now fail on duplicate bridges, there is no guarantee of this so we must check again. * The core bridges container allocation was changed to reject duplicate uniqueids instead of silently replacing an existing one. This is a "belt and suspenders" check. * A global mutex was added to bridge.c to prevent concurrent calls to `bridge_base_init()` and `bridge_register()`. * Even though you can no longer create multiple bridges with the same uniqueid at the same time, it's still possible that the bridge topic might be destroyed while a second bridge with the same uniqueid was trying to use it. To address this, the bridging core now increments the reference count on bridge->topic when a bridge is created and decrements it when the bridge is destroyed. * `bridge_create_common()` in res_stasis.c now checks the stasis app_bridges container to make sure a bridge with the requested uniqueid doesn't already exist. This may seem like overkill but there are so many entrypoints to bridge creation that we need to be safe and catch issues as soon in the process as possible. * The stasis app_bridges container allocation was changed to reject duplicate uniqueids instead of adding them. This is a "belt and suspenders" check. * The `bridge show all` CLI command now shows the bridge name as well as the bridge id. * Response code 409 "Conflict" was added as a possible response from the ARI bridge create resources to signal that a bridge with the requested uniqueid already exists. * Additional debugging was added to multiple bridging and stasis files. Resolves: #211 (cherry picked from commit 169b9b9a4f)	1 year ago