6998 Commits (d80869c4e439d8ffcab01870448dfa96bc5402c9)

Author	SHA1	Message	Date
Alexei Gradinari	b7dfe75744	build: remove pjsua, pjsystest, Python bindings and asterisk_malloc_debug stubs from pjproject dev build ... The pjsua Python module and the pjsua/pjsystest apps were used by the Asterisk Test Suite for SIP simulation in dev mode builds. They are now fully obsolete for three independent reasons: 1. **pjsua Python bindings officially deprecated upstream.** The pjproject maintainers added `pjsip-apps/src/python/DEPRECATED.txt` directing users to the PJSUA2 SWIG binding instead. A build-fix PR (https://github.com/pjsip/pjproject/pull/4892) was closed by the maintainer explicitly citing this deprecation. 2. **Removed from the Asterisk Test Suite.** As confirmed by @mbradeen: > *"We had to get rid of pjsua when we went to Python3 because it would > hang due to a conflict between async calls within pjsua and twisted. > There are still some old references to tests we couldn't fully convert > to sipp, but those are skipped."* 3. **Broken and unmaintained.** Building with Python 2.7 (the only version `configure.ac` searched for) fails with: ``` _pjsua.c: error: 'INIT_RETURN' undeclared (first use in this function) ``` due to a bug in pjproject 2.16's `python3_compat.h` that upstream declined to fix. This PR removes all pjsua-related build artifacts from Asterisk's bundled pjproject build: the pjsua and pjsystest application binaries, the deprecated Python (`_pjsua.so`) bindings, the `asterisk_malloc_debug.c` stubs, and the `PYTHONDEV` detection from `configure.ac`. Also removes `libpjsua` from Asterisk's main linker flags. DeveloperNote: The pjsua and pjsystest application binaries, the deprecated Python pjsua bindings (`_pjsua.so`), and the `asterisk_malloc_debug.c` stub implementations are no longer built or installed as part of the bundled pjproject dev mode build. The `PYTHONDEV` (python2.7-dev) build dependency is also removed. Developers who relied on the pjsua binary for Test Suite SIP simulation should use SIPp instead, which is the current Asterisk Test Suite standard. Fixes: #1840	3 months ago
Milan Kyselica	f9b6509bb0	callerid: fix signed char causing crash in MDMF parser ... Change rawdata buffer from char to unsigned char to prevent sign-extension of TLV length bytes >= 0x80. On signed-char platforms (all Asterisk builds due to -fsigned-char in configure.ac), these values become negative when assigned to int, bypass the `if (res > 32)` bounds check, and reach memcpy as size_t producing a ~18 EB read that immediately crashes with SIGSEGV. Affects DAHDI analog (FXO) channels only. Not reachable via SIP, PRI/BRI, or DTMF-based Caller ID. Fixes: #1839	3 months ago
Tinet-mucw	1ca3fb3936	pbx: Hold channel lock for exception datastore access ... ast_channel_datastore_find() and ast_channel_datastore_add() must only be called while the channel is locked (see channel.h). raise_exception() and the EXCEPTION dialplan function read path accessed the exception datastore without holding ast_channel_lock, which could corrupt the per-channel datastore list under concurrency and lead to crashes during teardown (e.g. double free in ast_datastore_free). Resolves: #1831	5 months ago
George Joseph	7dcf75cc74	xmldoc.c: Fix memory leaks in handling of provided_by. ... Added a few calls to ast_xml_free_attr() to squash memory leaks when handling "provided_by".	5 months ago
phoneben	f96ba7d11f	manager.c : Fix CLI event display ... manager.c: Fix CLI event display - `manager show events`: fix event names being truncated at 20 characters, widen column to 28 to accommodate the longest registered event name - `manager show events`: skip duplicate entries caused by multiple modules registering the same event name, list is already sorted so adjacent name comparison is sufficient	5 months ago
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
George Joseph	806b6ec4e9	rtp: Set RTPAUDIOQOS variables when ast_softhangup is called. ... If a channel in Stasis/ARI is hung up by the channel driver, the RTPAUDIOQOS variables are set before the channel leaves Stasis and are therefore available to the ARI app via ChannelVarset events. If the channel is hung up by ARI however, the channel leaves Stasis before the RTPAUDIOQOS variables are set so the app may not get the ChannelVarset events. We now set the RTPAUDIOQOS variables when ast_softhangup() is called as well as when the channel driver hangs up a channel. Since ARI hangups call ast_softhangup(), the variables will be set before the channel leaves Stasis and the app should get the ChannelVarset events. ast_rtp_instance_set_stats_vars(), which actually sets the variables, now checks to see if the variables are already set before attempting to set them. This prevents double messages from being generated. Resolves: #1816	5 months ago
Alexis Hadjisotiriou	7261968142	channel: Prevent crash during DTMF emulation when no timing module is loaded ... Description: When Asterisk is running without a timing module, attempting to process DTMF triggers a segmentation fault. This occurs because the system attempts to access a null timing file descriptor when setting up the DTMF emulation timer. This fix ensures that the system checks for a valid timing source before attempting to start the DTMF emulation timer. If no timing module is present, it logs a warning and skips the emulation instead of crashing the process. Changes: - Modified main/channel.c to add a safety check within the __ast_read function. - Implemented a graceful return path when no timing source is available - Added a LOG_WARNING to inform the administrator that DTMF emulation was skipped due to missing timing modules. Testing: - Disabled all timing_ modules in modules.conf and confirmed with 'timing test'. - Reproduced the crash by modifying the dialplan with: exten => 707,1,NoOp(Starting DTMF - No Timing Mode) same => n,Answer() same => n,Background(demo-congrats) same => n,WaitExten(10) same => n,Hangup() And calling 707 followed by 1 - Verified that with the fix applied, the system logs "No timing module loaded; skipping DTMF timer" and continues dialplan execution without crashing. - Confirmed stability during concurrent media sessions and DTMF input. Fixes: #566	5 months ago
Mike Bradeen	acfb214b3a	acl: Add ACL support to http and ari ... Add uri prefix based acl support to the built in http server. This allows an acl to be added per uri prefix (ie '/metrics' or '/ws') to restrict access. Add user based acl support for ARI. This adds new acl options to the user section of ari.conf to restrict access on a per user basis. resolves: #1799 UserNote: A new section, type=restriction has been added to http.conf to allow an uri prefix based acl to be configured. See http.conf.sample for examples and more information. The user section of ari.conf can now contain an acl configuration to restrict users access. See ari.conf.sample for examples and more information	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
Tinet-mucw	ac53fba401	main/file: fix translated-frame write loop to use current frame ... write each translated frame from translator output. Resolves: #1797	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
Naveen Albert	11393d511e	build: Fix unused-but-set-variable warnings with gcc 16. ... Fix or remove a few variables that were being set but not actually used anywhere, causing warnings with gcc 16. Resolves: #1783	5 months ago
Naveen Albert	90d9baf1b2	build: Fix another GCC discarded-qualifiers const error. ... Follow on commit to 27a39cba7e to fix compilation with BETTER_BACKTRACES with gcc 15.2.1. Resolves: #1781	5 months ago
Joshua C. Colp	c3af03e823	build: Fix GCC discarded-qualifiers const errors. ... GCC 15.2.1 pays attention to the discarding of the const qualifier when strchr, strrchr, memchr, or memrchr are now used. This change fixes numerous errors with this throughout the tree. The fixes can be broken down into the following: 1. The return value should be considered const. 2. The value passed to strchr or strrchr can be cast as it is expected and allowed to be modified. 3. The pointer passed to strchr or strrchr is not meant to be modified and so the contents must be duplicated. 4. It was declared const and never should have been.	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
phoneben	df3d94d805	stasis.c: Fix deadlock in stasis_topic_pool_get_topic during module load ... stasis.c: Fix deadlock in stasis_topic_pool_get_topic during module load. Deadlock occurs when res_manager_devicestate loads concurrently with device state operations due to lock ordering violation: Thread 1: Holds pool lock → needs topic lock (in stasis_forward_all) Thread 2: Holds topic lock → needs pool lock (in stasis_topic_pool_get_topic) Fix: Release pool lock before calling stasis_topic_create() and stasis_forward_all(). Re-acquire only for insertion with race check. Preserves borrowed reference semantics while breaking the deadlock cycle. Fixes: #1611	5 months ago
Jasper Hafkenscheid	257de0c05a	channelstorage_cpp_map_name_id: Fix get_by_name_prefix prefix match ... Lower bound filter did not ensure prefix match. Resolves: #1730	5 months ago
Talha Asghar	6722bfa152	say.c: added language support for pashto and dari ... With this new feature, users who speak these languages can now benefit from the text-to-speech functionality provided by asterisk. This will make the platform more accessible and useful to a wider range of users, particularly those in regions where Pashto and Dari are spoken. This contribution will help to improve the overall usability and inclusivity of the asterisk platform. Fixes: #1724	5 months ago
George Joseph	2452fd343b	xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile. ... The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external entities, could allow a potential XXE injection attack. Replacing it with XML_PARSE_NONET, which prevents network access, is safer. Resolves: #GHSA-85x7-54wr-vh42	7 months ago
George Joseph	012ec29059	http.c: Change httpstatus to default disabled and sanitize output. ... To address potential security issues, the httpstatus page is now disabled by default and the echoed query string and cookie output is html-escaped. Resolves: #GHSA-v6hp-wh3r-cwxh UpgradeNote: To prevent possible security issues, the `/httpstatus` page served by the internal web server is now disabled by default. To explicitly enable it, set `enable_status=yes` in http.conf.	7 months ago
Sean Bright	c738d23ee5	asterisk.c: Use C.UTF-8 locale instead of relying on user's environment. ... Resolves: #1739	7 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.	7 months ago
Sean Bright	432be37a58	asterisk.c: Allow multi-byte characters on the Asterisk CLI. ... Versions of libedit that support Unicode expect that the EL_GETCFN (the function that does character I/O) will fill in a `wchar_t` with a character, which may be multi-byte. The built-in function that libedit provides, but does not expose with a public API, does properly handle multi-byte sequences. Due to the design of Asterisk's console processing loop, Asterisk provides its own implementation which does not handle multi-byte characters. Changing Asterisk to use libedit's built-in function would be ideal, but would also require changing some fundamental things about console processing which could be fairly disruptive. Instead, we bring in libedit's `read_char` implementation and modify it to suit our specific needs. Resolves: #60	7 months ago
sarangr7	a1af0b56f1	main/dial.c: Set channel hangup cause on timeout in handle_timeout_trip ... When dial attempts timeout in the core dialing API, the channel's hangup cause was not being set before hanging up. Only the ast_dial_channel structure's internal cause field was updated, but the actual ast_channel hangup cause remained unset. This resulted in incorrect or missing hangup cause information being reported through CDRs, AMI events, and other mechanisms that read the channel's hangup cause when dial timeouts occurred via applications using the dialing API (FollowMe, Page, etc.). The fix adds proper channel locking and sets AST_CAUSE_NO_ANSWER on the channel before calling ast_hangup(), ensuring consistent hangup cause reporting across all interfaces. Resolves: #1660	7 months ago
Sean Bright	132e110647	http.c: Include remote address in URI handler message. ... Resolves: #1662	7 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.	7 months ago
George Joseph	56aafb3113	endpoint.c: Plug a memory leak in ast_endpoint_shutdown(). ... Commit 26795be introduced a memory leak of ast_endpoint when ast_endpoint_shutdown() was called. The leak occurs only if a configuration change removes an endpoint and isn't related to call volume or the length of time asterisk has been running. An ao2_ref(-1) has been added to ast_endpoint_shutdown() to plug the leak. Resolves: #1635	7 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	7 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>	7 months ago
Michal Hajek	1b98a78cd6	manager: fix double free of criteria variable when adding filter ... Signed-off-by: Michal Hajek <michal.hajek@daktela.com> Fixes: #1531	7 months ago
Sean Bright	6dcad1242a	abstract_jb.c: Remove redundant timer check per static analysis. ... While this check is technically unnecessary, it also was not harmful. The 2 other items mentioned in the linked issue are false positives and require no action. Resolves: #1417	7 months ago
phoneben	3e14d0cf9b	channelstorage_cpp: Fix fallback return value in channelstorage callback ... callback returned the last iterated channel when no match existed, causing invalid channel references and potential double frees. Updated to correctly return NULL when there is no match. Resolves: #1609	7 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.	7 months ago
Naveen Albert	5354e8a001	pbx.c: Print new context count when reloading dialplan. ... When running "dialplan reload", the number of contexts reported is initially wrong, as it is the old context count. Running "dialplan reload" a second time returns the correct number of contexts that are loaded. This can confuse users into thinking that the reload didn't work successfully the first time. This counter is currently only incremented when iterating the old contexts prior to the context merge; at the very end, get the current number of elements in the context hash table and report that instead. This way, the count is correct immediately whenever a reload occurs. Resolves: #1599	7 months ago
Tinet-mucw	396a4f153f	core_unreal.c: Use ast instead of p->chan to get the DIALSTATUS variable ... After p->chan = NULL, ast still points to the valid channel object, using ast safely accesses the channel's DIALSTATUS variable before it's fully destroyed Resolves: #1590	7 months ago
phoneben	390ca4e827	stasis: switch stasis show topics temporary container from list - RBtree ... switch stasis show topics temporary container from list to RB-tree minimizing lock time Resolves: #1585	7 months ago
Sean Bright	3238e4e495	main: Explicitly mark case statement fallthrough as such. ... Resolves: #1442	7 months ago
George Joseph	2c75447235	channelstorage: Allow storage driver read locking to be skipped. ... After PR #1498 added read locking to channelstorage_cpp_map_name_id, if ARI channels/externalMedia was called with a custom channel id AND the cpp_map_name_id channel storage backend is in use, a deadlock can occur when hanging up the channel. It's actually triggered in channel.c:__ast_channel_alloc_ap() when it gets a write lock on the channelstorage driver then subsequently does a lookup for channel uniqueid which now does a read lock. This is an invalid operation and causes the lock state to get "bad". When the channels try to hang up, a write lock is attempted again which hangs and causes the deadlock. Now instead of the cpp_map_name_id channelstorage driver "get" APIs automatically performing a read lock, they take a "lock" parameter which allows a caller who already has a write lock to indicate that the "get" API must not attempt its own lock. This prevents the state from getting mesed up. The ao2_legacy driver uses the ao2 container's recursive mutex so doesn't have this issue but since it also implements the common channelstorage API, it needed its "get" implementations updated to take the lock parameter. They just don't use it. Resolves: #1578	10 months ago
Joshua C. Colp	54da3e1b9e	Revert "pjsip: Move from threadpool to taskpool" ... This reverts commit bb6b76c2d8.	10 months ago
Tinet-mucw	38f2fd2527	iostream.c: Handle TLS handshake attacks in order to resolve the issue of exceeding the maximum number of HTTPS sessions. ... The TCP three-way handshake completes, but if the server is under a TLS handshake attack, asterisk will get stuck at SSL_do_handshake(). In this case, a timeout mechanism should be set for the SSL/TLS handshake process to prevent indefinite waiting during the SSL handshake. Resolves: #1559	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
Joshua C. Colp	b6556cc789	devicestate: Don't publish redundant device state messages. ... When publishing device state check the local cache for the existing device state. If the new device state is unchanged from the prior one, don't bother publishing the update. This can reduce the work done by consumers of device state, such as hints and app_queue, by not publishing a message to them. These messages would most often occur with devices that are seeing numerous simultaneous channels. The underlying device state would remain as in use throughout, but an update would be published as channels are created and hung up.	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
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
Naveen Albert	9a4f6f4c4a	core_unreal: Preserve ADSI capability when dialing Local channels. ... Dial() already preserves the ADSI capability by copying it to the new channel, but since Local channel pairs consist of two channels, we also need to copy the capability to the second channel. Resolves: #1517	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