7017 Commits (13de0495f29f9997b96b09f7a5357e1fb2a87fdf)

Author	SHA1	Message	Date
Alexander Bakker	8be216f74e	abstract_jb.c: Remove timerfd from channel when disabling jitter buffer ... Previously, the lingering timerfd would cause a tight loop if the channel enters a BridgeWait after the jitter buffer was disabled. Fixes: #1762	4 months ago
George Joseph	0aa6b2c0c4	Ensure channel locks aren't held while calling ast_set_variables. ... If the channel is locked when calling ast_set_variables and any of the variables contained dialplan functions, there's a possiblilty of a deadlock. To prevent this, either the explicit locks were removed or the call to ast_set_variables moved out of the lock scope. A warning to not hold channel locks is also added to the documentation for ast_set_variables. Resolves: #1936	4 months ago
George Joseph	2fedb5d195	channel.c: Move setting RTP stats from ast_softhangup to ast_ari_channels_hangup. ... The original trigger for setting the RTP stats in ast_softhangup() came from an ARI issue where stats weren't being set in time to be reported on STASIS_END events. The thought was that setting them in a common place like ast_softhangup() would ensure the stats were set in possibly other scenarios. Unfortunately, setting the RTP stats variables in ast_softhangup() broke ABI as it required that no channel locks be held which was not the case earlier. Given that the original issue was ARI, we can move setting the stats to ast_ari_channels_hangup() in resource_channels just before it calls ast_softhangup(). This might not catch all cases of the stats not being set, but it won't break ABI or deadlock either. Resolves: #1928	4 months ago
George Joseph	deb8a51a26	res_rtp_asterisk: Add option to control stun host resolution when TTL = 0 ... If a hostname is specified for stunaddr in rtp.conf, periodic DNS resolution is enabled based on the TTL returned in the DNS results. If the TTL returned is 0, it means that the next time the IP address is needed, it must be looked up again. I.E. Don't cache. Historically (and incorrectly) however, res_rtp_asterisk stopped the periodic resolution and never re-resolved the hostname again. Besides what's mentioned in the user notes... * Additional debugging was added in various STUN/DNS functions. * The `rtp show settings` CLI command shows more detailed STUN info. * Some debugging was added to dns_core.c and dns_recurring.c. UserNote: A new `stunaddr_reresolve_ttl_0` parameter has been added to rtp.conf that allows control over what happens when a STUN server hostname lookup returns a TTL of 0. The values can be set as follows: - 'no': This is the historical (and current default) behavior of not doing any further lookups and continuing to use the last successful result until Asterisk is restarted or rtp.conf is reloaded. - 'yes': Use the last cached result for the current call but trigger re-resolution in the background for the benefit of future calls. If the result of the background lookup is a ttl > 0, periodic resolution will be restarted otherwise the next call will use the new cached value and will trigger a background lookup again. UserNote: A new CLI command `rtp resolve stun hostname` has been added that will force a resolution of the STUN hostname and (re)start periodic resolution if the result has a TTL > 0. Resolves: #1858	4 months ago
Joshua C. Colp	7c85f7dd9e	manager: Eliminate unnecessary code, simplify sessions in stasis callbacks ... Due to stasis filtering the stasis callback for AMI type messages is guaranteed to only receive messages that can be turned into AMI events, so remove the check done in the callback. The sessions container usage for the stasis callbacks has also been simplified by having a reference on the message router subscription instead of having to acquire the sessions from the global object each time.	5 months ago
George Joseph	6e23ff15ef	channel.c: Don't lock the channel in ast_softhangup while setting rtp instance vars ... ast_softhangup() was locking the channel before calling ast_rtp_instance_set_stats_vars() which, if the channel was in a bridge, then locked the bridge peer channel. If another thread attempted to set bridge variables on the peer, it would lock that channel first, then this channel causing a lock inversion. ast_softhangup() now holds the channel lock while retrieving the rtp instance, then unlocks it before calling ast_rtp_instance_set_stats_vars(), then locks it again after it returns. Resolves: #1907	5 months ago
mattia	8e221dfd03	res_pjsip: Add per-endpoint RTP port range configuration ... Add rtp_port_start and rtp_port_end options to PJSIP endpoint configuration, allowing each endpoint to use a dedicated RTP port range instead of the global rtp.conf setting. This is useful for scenarios where different endpoints need isolated port ranges, such as firewall rules per trunk, multi-tenant systems, or network QoS policies tied to port ranges. The implementation adds ast_rtp_instance_new_with_port_range() to the RTP engine API, which sets the port range on the instance before the engine allocates the transport. The default RTP engine (res_rtp_asterisk) checks for per-instance overrides in rtp_allocate_transport() and falls back to the global range when none is set. Both options must be set together, with values >= 1024 and rtp_port_end > rtp_port_start. Setting both to 0 (the default) preserves existing behavior. Resolves: https://github.com/asterisk/asterisk-feature-requests/issues/71 UserNote: PJSIP endpoints now support rtp_port_start and rtp_port_end options to configure a dedicated RTP port range per endpoint, overriding the global rtp.conf setting. UpgradeNote: An alembic database migration has been added to add the rtp_port_start and rtp_port_end columns to the ps_endpoints table. Run "alembic upgrade head" to apply the schema change. DeveloperNote: New public API: ast_rtp_instance_new_with_port_range() creates an RTP instance with a per-instance port range. ast_rtp_instance_get_port_start() and ast_rtp_instance_get_port_end() allow RTP engines to query the override. Third-party RTP engines can use these getters to support per-instance port ranges.	5 months ago
George Joseph	227795eace	asterisk.c: Fix #if HAVE_LIBEDIT_IS_UNICODE. ... Line 2729 has `#if HAVE_LIBEDIT_IS_UNICODE` instead if `#ifdef`. Since macros defined by autoconf are either set to `1` or not set at all, older distros where libedit isn't unicode won't have that macro defined and will fail to compile. Resolves: #1896	5 months ago
Sebastian Jennen	861f5b1216	translate.c: implement different sample_types for translation computation. ... The default (codec) still uses the codec provided samples. Additionally different sample_types can be used with eg: `translate sampletype speech` and then running `core show translation comp 10` to measure performance of different audio scenarios. Resolves: #1807	5 months ago
George Joseph	c45232f02b	pbx_functions: Save module pointer before calling read and write callbacks. ... Before ast_func_read and ast_func_write call their respective read and write callbacks for registered dialplan functions, they use the module pointer in the registered ast_custom_function structure to increment the module use count. They then decrement the usecount when the callback returns. This prevents the providing module from being unloaded while there's a call using the function. Some modules, notably func_odbc, create and destroy dialplan functions based on the contents of a config file. Since the ast_custom_function structure is dynamically allocated, it could be destroyed on reload which means when the module's read or write callback returns to the ast_func calls it would try to decrement the usecount using the module pointer from an ast_custom_function structure that has already been freed. Proper locking or reference counting by the module can reduce the possibility of this happening but it can't prevent it because it doesn't have control after its read or write callback has returned to ast_func_read or ast_func_write. To address this, ast_func_read, ast_func_read2 and ast_func_write save the module pointer to a local variable before calling the module's callback, then use the saved pointer to decrement the use count. The module pointer will always be valid if the module is loaded regardless of the state of the ast_custom_function structure. Resolves: #1818	5 months ago
Alexei Gradinari	43cda0d8d2	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	6 months ago
Milan Kyselica	15bbe5c58d	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	6 months ago
Tinet-mucw	ceceea4b38	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	6 months ago
George Joseph	0c25c9c118	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".	6 months ago
phoneben	a31a901a44	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	7 months ago
George Joseph	041a183bac	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	7 months ago
George Joseph	836696ad06	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	7 months ago
Alexis Hadjisotiriou	04edbe2bc6	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	7 months ago
Mike Bradeen	ffec03d8e8	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	7 months ago
Naveen Albert	9f25e3543a	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	7 months ago
Tinet-mucw	be0ffa1153	main/file: fix translated-frame write loop to use current frame ... write each translated frame from translator output. Resolves: #1797	7 months ago
George Joseph	dcedc2dc0e	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.	7 months ago
George Joseph	4871413877	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.	7 months ago
Prashant Srivastav	2f54f5b19a	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)	7 months ago
Naveen Albert	ad7be5c675	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	7 months ago
Naveen Albert	492d5e87a5	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	7 months ago
Joshua C. Colp	7aba01a49d	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.	7 months ago
Joshua C. Colp	af3ca1b6cd	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.	7 months ago
George Joseph	27f35ca2fd	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.	8 months ago
George Joseph	2da33e230e	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	8 months ago
phoneben	9dfd01d53c	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	8 months ago
Sean Bright	19851fdf59	asterisk.c: Use C.UTF-8 locale instead of relying on user's environment. ... Resolves: #1739	8 months ago
Jasper Hafkenscheid	da8814524b	channelstorage_cpp_map_name_id: Fix get_by_name_prefix prefix match ... Lower bound filter did not ensure prefix match. Resolves: #1730	8 months ago
Talha Asghar	0a76b99594	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	8 months ago
Sean Bright	fb433fdba6	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.	9 months ago
Sean Bright	7088b5c661	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	9 months ago
sarangr7	c37d2cd0d7	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	9 months ago
Sean Bright	4dd0537a0f	http.c: Include remote address in URI handler message. ... Resolves: #1662	9 months ago
Joshua C. Colp	a12251ca88	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
George Joseph	3fe9377c90	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	10 months ago
Sean Bright	786156e3ba	Revert "func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()" ... This reverts commit 5177662990. For rationale, see #1621 and #1606	10 months ago
Mike Bradeen	5f2c9ffd0d	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>	10 months ago
Michal Hajek	3bd4852fa6	manager: fix double free of criteria variable when adding filter ... Signed-off-by: Michal Hajek <michal.hajek@daktela.com> Fixes: #1531	10 months ago
Sean Bright	abcf456715	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	10 months ago
phoneben	8dec33f963	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	10 months ago
George Joseph	724f4cb6e8	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.	10 months ago
Naveen Albert	d4ff1a10f4	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	10 months ago
Tinet-mucw	c1d8dead62	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	11 months ago
phoneben	0361a8e9d7	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	11 months ago
Sean Bright	b64bd60df3	main: Explicitly mark case statement fallthrough as such. ... Resolves: #1442	11 months ago