asterisk

Commit Graph

Author	SHA1	Message	Date
Sven Kube	30daba71bc	stasis_channels.c: Add null check for referred_by in ast_ari_transfer_message_create When handling SIP transfers via ARI, the `referred_by` field in `transfer_ari_state` may be null, since SIP REFER requests are not required to include a `Referred-By` header. Without this check, a null value caused the transfer to fail and triggered a NOTIFY with a 500 Internal Server Error.	12 months ago
Joshua C. Colp	e4aa5bbd22	taskpool: Update versions for taskpool stasis options.	12 months ago
Joshua C. Colp	0c49ba4478	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.	12 months ago
Nathan Monfils	1657ccfc68	manager.c: Fix presencestate object leak ast_presence_state allocates subtype and message. We straightforwardly need to clean those up.	1 year ago
Sean Bright	d274905fa5	audiohook.c: Ensure correct AO2 reference is dereffed. Part of #1440.	1 year ago
Ben Ford	a85cbadf83	rtp_engine.c: Add exception for comfort noise payload. In a previous commit, a change was made to ast_rtp_codecs_payload_code_tx_sample_rate to check for differing sample rates. This ended up returning an invalid payload int for comfort noise. A check has been added that returns early if the payload is in fact supposed to be comfort noise. Fixes: #1340	1 year ago
Naveen Albert	d9cbadc51b	pbx_variables.c: Create real channel for "dialplan eval function". "dialplan eval function" has been using a dummy channel for function evaluation, much like many of the unit tests. However, sometimes, this can cause issues for functions that are not expecting dummy channels. As an example, ast_channel_tech(chan) is NULL on such channels, and ast_channel_tech(chan)->type consequently results in a NULL dereference. Normally, functions do not worry about this since channels executing dialplan aren't dummy channels. While some functions are better about checking for these sorts of edge cases, use a real channel with a dummy technology to make this CLI command inherently safe for any dialplan function that could be evaluated from the CLI. Resolves: #1434	1 year ago
Naveen Albert	171d5e996c	pbx_builtins: Allow custom tone for WaitExten. Currently, the 'd' option will play dial tone while waiting for digits. Allow it to accept an argument for any tone from indications.conf. Resolves: #1396 UserNote: The tone used while waiting for digits in WaitExten can now be overridden by specifying an argument for the 'd' option.	1 year ago
Alexei Gradinari	14ea4d3b39	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.	1 year ago
George Joseph	5d822d64ef	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.	1 year ago
Naveen Albert	dd9c71ed0b	dsp.c: Improve debug logging in tone_detect(). The debug logging during DSP processing has always been kind of overwhelming and annoying to troubleshoot. Simplify and improve the logging in a few ways to aid DSP debugging: * If we had a DSP hit, don't also emit the previous debug message that was always logged. It is duplicated by the hit message, so this can reduce the number of debug messages during detection by 50%. * Include the hit count and required number of hits in the message so on partial detections can be more easily troubleshot. * Use debug level 9 for hits instead of 10, so we can focus on hits without all the noise from the per-frame debug message. * 1-index the hit count in the debug messages. On the first hit, it currently logs '0', just as when we are not detecting anything, which can be confusing. Resolves: #1375	1 year ago
George Joseph	59c9aa00b0	xmldoc.c: Fix rendering of CLI output. If you do a `core show application Dial`, you'll see it's kind of a mess. Indents are wrong is some places, examples are printed in black which makes them invisible on most terminals, and the lack of line breaks in some cases makes it hard to follow. * Fixed the rendering of examples so they are indented properly and changed the color so they can be seen. * There is now a line break before each option. * Options are now printed on their own line with all option content indented below them. Example from Dial before fixes: ``` Example: Dial 555-1212 on first available channel in group 1, searching from highest to lowest Example: Ringing FXS channel 4 with ring cadence 2 Example: Dial 555-1212 on channel 3 and require answer confirmation ... O([mode]): mode - With <mode> either not specified or set to '1', the originator hanging up will cause the phone to ring back immediately. - With <mode> set to '2', when the operator flashes the trunk, it will ring their phone back. Enables operator services mode. This option only works when bridging a DAHDI channel to another DAHDI channel only. If specified on non-DAHDI interfaces, it will be ignored. When the destination answers (presumably an operator services station), the originator no longer has control of their line. They may hang up, but the switch will not release their line until the destination party (the operator) hangs up. p: This option enables screening mode. This is basically Privacy mode without memory. ``` After: ``` Example: Dial 555-1212 on first available channel in group 1, searching from highest to lowest same => n,Dial(DAHDI/g1/5551212) Example: Ringing FXS channel 4 with ring cadence 2 same => n,Dial(DAHDI/4r2) Example: Dial 555-1212 on channel 3 and require answer confirmation same => n,Dial(DAHDI/3c/5551212) ... O([mode]): mode - With <mode> either not specified or set to '1', the originator hanging up will cause the phone to ring back immediately. With <mode> set to '2', when the operator flashes the trunk, it will ring their phone back. Enables operator services mode. This option only works when bridging a DAHDI channel to another DAHDI channel only. If specified on non-DAHDI interfaces, it will be ignored. When the destination answers (presumably an operator services station), the originator no longer has control of their line. They may hang up, but the switch will not release their line until the destination party (the operator) hangs up. p: This option enables screening mode. This is basically Privacy mode without memory. ``` There are still things we can do to make this more readable but this is a start.	1 year ago
Naveen Albert	1ddf63d83f	func_frame_drop: Add debug messages for dropped frames. Add debug messages in scenarios where frames that are usually processed are dropped or skipped. Resolves: #1371	1 year ago
Naveen Albert	a23c467c82	bridge.c: Obey BRIDGE_NOANSWER variable to skip answering channel. If the BRIDGE_NOANSWER variable is set on a channel, it is not supposed to answer when another channel bridges to it using Bridge(), and this is checked when ast_bridge_call* is called. However, another path exists (bridge_exec -> ast_bridge_add_channel) where this variable was not checked and channels would be answered. We now check the variable there. Resolves: #401 Resolves: #1364	1 year ago
Allan Nathanson	31571a5079	file.c: with "sounds_search_custom_dir = yes", search "custom" directory With `sounds_search_custom_dir = yes`, we are supposed to search for sounds in the `AST_DATA_DIR/sounds/custom` directory before searching the normal directories. Unfortunately, a recent change (https://github.com/asterisk/asterisk/pull/1172) had a typo resulting in the "custom" directory not being searched. This change restores this expected behavior. Resolves: #1353	1 year ago
Sperl Viktor	c1f24b74d7	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.	1 year ago
Naveen Albert	c8f844a900	logger.c: Remove deprecated/redundant configuration option. Remove the deprecated 'rotatetimestamp' config option, as this was deprecated by 'rotatestrategy' in 1.6 by commit `f5a14167f3`. Resolves: #1345 UpgradeNote: The deprecated rotatetimestamp option has been removed. Use rotatestrategy instead.	1 year ago
Naveen Albert	5c3cd44563	cli.c: Remove deprecated and redundant CLI command. The "no debug channel" command has been deprecated since 1.6 (commit `691363656f`), as it is replaced by "core set debug channel", which also supports tab-completion on channels. Remove the redundant command. Resolves: #1343 UpgradeNote: The deprecated "no debug channel" command has now been removed; use "core set debug channel" instead.	1 year ago
George Joseph	baa73b6b12	channelstorage_cpp_map_name_id.cc: Refactor iterators for thread-safety. The fact that deleting an object from a map invalidates any iterator that happens to currently point to that object was overlooked in the initial implementation. Unfortunately, there's no way to detect that an iterator has been invalidated so the result was an occasional SEGV triggered by modules like app_chanspy that opens an iterator and can keep it open for a long period of time. The new implementation doesn't keep the underlying C++ iterator open across calls to ast_channel_iterator_next() and uses a read lock on the map to ensure that, even for the few microseconds we use the iterator, another thread can't delete a channel from under it. Even with this change, the iterators are still WAY faster than the ao2_legacy storage driver. Full details about the new implementation are located in the comments for iterator_next() in channelstorage_cpp_map_name_id.cc. Resolves: #1309	1 year ago
zhou_jiajian	511d22ef5e	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'	1 year ago
George Joseph	43bf8a4ded	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.	1 year ago
Alexei Gradinari	3e178dcfd6	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	1 year ago
Tinet-mucw	009e3ef3f5	pbx.c: When the AST_SOFTHANGUP_ASYNCGOTO flag is set, pbx_extension_helper should return directly. 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 pbx_extension_helper this info is changed. This will cause the current dialplan location to be executed twice. In other words, the Redirect action does not take effect. Resolves: #1315	1 year ago
George Joseph	32b8bf0169	cdr.c: Set tenantid from party_a->base instead of chan->base. The CDR tenantid was being set in cdr_object_alloc from the channel->base snapshot. Since this happens at channel creation before the dialplan is even reached, calls to `CHANNEL(tenantid)=<something>` in the dialplan were being ignored. Instead we now take tenantid from party_a when cdr_object_create_public_records() is called which is after the call has ended and all channel snapshots rebuilt. This is exactly how accountcode and amaflags, which can also be set in tha dialplpan, are handled. Resolves: #1259	1 year ago
George Joseph	0086f4a0b8	channelstorage_cpp_map_name_id: Fix callback returning non-matching channels. When the callback() API was invoked but no channel passed the test, callback would return the last channel tested instead of NULL. It now correctly returns NULL when no channel matches. Resolves: #1288	1 year ago
Naveen Albert	415daae95f	users.conf: Remove deprecated users.conf integration. users.conf was deprecated in Asterisk 21 and is now being removed for Asterisk 23, in accordance with the Asterisk deprecation policy. This consists of: * Removing integration with app_directory, app_voicemail, chan_dahdi, chan_iax2, and AMI. * users.conf was also partially used for res_phoneprov, and this remaining functionality is consolidated to a separate phoneprov_users.conf, used only by res_phoneprov. Resolves: #1292 UpgradeNote: users.conf has been removed and all channel drivers must be configured using their specific configuration files. The functionality previously in users.conf for res_phoneprov is now in phoneprov_users.conf.	1 year ago
George Joseph	5963e624e2	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.	1 year ago
George Joseph	d5bd2b3ce9	channelstorage: Rename callbacks that conflict with DEBUG_FD_LEAKS. DEBUG_FD_LEAKS replaces calls to "open" and "close" with functions that keep track of file descriptors, even when those calls are actually callbacks defined in structures like ast_channelstorage_instance->open and don't touch file descriptors. This causes compilation failures. Those callbacks have been renamed to "open_instance" and "close_instance" respectively. Resolves: #1287	1 year ago
Michal Hajek	4522eb1222	audiohook.c: Improve frame pairing logic to avoid MixMonitor breakage with mixed codecs This patch adjusts the read/write synchronization logic in audiohook_read_frame_both() to better handle calls where participants use different codecs or sample sizes (e.g., alaw vs G.722). The previous hard threshold of 2 * samples caused MixMonitor recordings to break or stutter when frames were not aligned between both directions. The new logic uses a more tolerant limit (1.5 * samples), which prevents audio tearing without causing excessive buffer overruns. This fix specifically addresses issues with MixMonitor when recording directly on a channel in a bridge using mixed codecs. Reported-by: Michal Hajek <michal.hajek@daktela.com> Resolves: #1276 Resolves: #1279	1 year ago
Sean Bright	574ddb9eae	channelstorage_makeopts.xml: Remove errant XML character. Resolves: #1282	1 year ago
Tinet-mucw	8149554e90	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	1 year ago
George Joseph	c873f2ae7e	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	1 year ago
George Joseph	3a5ffe2842	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	1 year ago
mkmer	abf3f78c81	frame.c: validate frame data length is less than samples when adjusting volume Resolves: #1230	1 year ago
Nathan Monfils	7805f2892a	manager.c: Invalid ref-counting when purging events We have a use-case where we generate a lot of events on the AMI, and then when doing `manager show eventq` we would see some events which would linger for hours or days in there. Obviously something was leaking. Testing allowed us to track down this logic bug in the ref-counting on the event purge. Reproducing the bug was not super trivial, we managed to do it in a production-like load testing environment with multiple AMI consumers. The race condition itself: 1. something allocates and links `session` 2. `purge_sessions` iterates over that `session` (takes ref) 3. `purge_session` correctly de-referencess that session 4. `purge_session` re-evaluates the while() loop, taking a reference 5. `purge_session` exits (`n_max > 0` is false) 6. whatever allocated the `session` deallocates it, but a reference is now lost since we exited the `while` loop before de-referencing. 7. since the destructor is never called, the session->last_ev->usecount is never decremented, leading to events lingering in the queue The impact of this bug does not seem major. The events are small and do not seem, from our testing, to be causing meaningful additional CPU usage. Mainly we wanted to fix this issue because we are internally adding prometheus metrics to the eventq and those leaked events were causing the metrics to show garbage data.	1 year ago
George Joseph	8f1982c4d6	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	1 year ago
Naveen Albert	2ced79259a	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.	1 year ago
George Joseph	f302c116b4	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().	1 year ago
Peter Jannesen	032584115b	action_redirect: remove after_bridge_goto_info Under certain circumstances the context/extens/prio are stored in the after_bridge_goto_info. This info is used when the bridge is broken by for hangup of the other party. In the situation that the bridge is broken by an AMI Redirect this info is not used but also not removed. With the result that when the channel is put back in a bridge and the bridge is broken the execution continues at the wrong context/extens/prio. Resolves: #1144	1 year ago
Joshua C. Colp	66c01d8b22	channel: Always provide cause code in ChannelHangupRequest. When queueing a channel to be hung up a cause code can be specified in one of two ways: 1. ast_queue_hangup_with_cause This function takes in a cause code and queues it as part of the hangup request, which ultimately results in it being set on the channel. 2. ast_channel_hangupcause_set + ast_queue_hangup This combination sets the hangup cause on the channel before queueing the hangup instead of as part of that process. In the #2 case the ChannelHangupRequest event would not contain the cause code. For consistency if a cause code has been set on the channel it will now be added to the event. Resolves: #1197	1 year ago
George Joseph	c52136c277	asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf. Commands in the "[startup_commands]" section of cli.conf have historically run after all core and module initialization has been completed and just before "Asterisk Ready" is printed on the console. This meant that if you wanted to debug initialization of a specific module, your only option was to turn on debug for everything by setting "debug" in asterisk.conf. This commit introduces options to allow you to run CLI commands earlier in the asterisk startup process. A command with a value of "pre-init" will run just after logger initialization but before most core, and all module, initialization. A command with a value of "pre-module" will run just after all core initialization but before all module initialization. A command with a value of "fully-booted" (or "yes" for backwards compatibility) will run as they always have been...after all initialization and just before "Asterisk Ready" is printed on the console. This means you could do this... ``` [startup_commands] core set debug 3 res_pjsip.so = pre-module core set debug 0 res_pjsip.so = fully-booted ``` This would turn debugging on for res_pjsip.so to catch any module initialization debug messages then turn it off again after the module is loaded. UserNote: In cli.conf, you can now define startup commands that run before core initialization and before module initialization.	1 year ago
Sean Bright	9fdf8597cb	channel: Deprecate `ast_moh_cleanup(...)`. We don't want anyone calling it but the channel destructor.	1 year ago
George Joseph	3d5ae0b5e1	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/	1 year ago
mkmer	edadca7151	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	1 year ago
Allan Nathanson	113c1af4f3	file.c: missing "custom" sound files should not generate warning logs With `sounds_search_custom_dir = yes` we first look to see if a sound file is present in the "custom" sound directory before looking in the standard sound directories. We should not be issuing a WARNING log message if a sound cannot be found in the "custom" directory. Resolves: https://github.com/asterisk/asterisk/issues/1170	1 year ago
Ben Ford	f6d2962566	documentation: Update Gosub, Goto, and add new documentationtype. Gosub and Goto were not displaying their syntax correctly on the docs site. This change adds a new way to specify an optional context, an optional extension, and a required priority that the xml stylesheet can parse without having to know which optional parameters come in which order. In Asterisk, it looks like this: parameter name="context" documentationtype="dialplan_context" parameter name="extension" documentationtype="dialplan_extension" parameter name="priority" documentationtype="dialplan_priority" required="true" The stylesheet will ignore the context and extension parameters, but for priority, it will automatically inject the following: [[context,]extension,]priority This is the correct oder for applications such as Gosub and Goto.	1 year ago
Naveen Albert	f0f220b369	config.c: Fix inconsistent pointer logic in ast_variable_update. Commit `3cab4e7ab4` introduced a regression by causing the wrong pointers to be used in certain (more complex) cases. We now take care to ensure the exact same pointers are used as before that commit, and simplify by eliminating the unnecessary second for loop. Resolves: #1147	2 years ago
Sean Bright	c7db162844	docs: AMI documentation fixes. Most of this patch is adding missing PJSIP-related event documentation, but the one functional change was adding a sorcery to-string handler for endpoint's `redirect_method` which was not showing up in the AMI event details or `pjsip show endpoint <endpoint>` output. The rest of the changes are summarized below: * app_agent_pool.c: Typo fix Epoche -> Epoch. * stasis_bridges.c: Add missing AttendedTransfer properties. * stasis_channels.c: Add missing AgentLogoff properties. * pjsip_manager.xml: - Add missing AorList properties. - Add missing AorDetail properties. - Add missing ContactList properties. - Add missing ContactStatusDetail properties. - Add missing EventDetail properties. - Add missing AuthList properties. - Add missing AuthDetail properties. - Add missing TransportDetail properties. - Add missing EndpointList properties. - Add missing IdentifyDetail properties. * res_pjsip_registrar.c: Add missing InboundRegistrationDetail documentation. * res_pjsip_pubsub.c: - Add missing ResourceListDetail documentation. - Add missing InboundSubscriptionDetail documentation. - Add missing OutboundSubscriptionDetail documentation. * res_pjsip_outbound_registration.c: Add missing OutboundRegistrationDetail documentation.	2 years ago
Allan Nathanson	9277311f93	config.c: #include of non-existent file should not crash Corrects a segmentation fault when a configuration file has a #include statement that referenced a file that does not exist. Resolves: https://github.com/asterisk/asterisk/issues/1139	2 years ago
George Joseph	1e469a5e06	manager.c: Check for restricted file in action_createconfig. The `CreateConfig` manager action now ensures that a config file can only be created in the AST_CONFIG_DIR unless `live_dangerously` is set. Resolves: #1122	2 years ago

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6949 Commits (30daba71bc231db3ecc71619b6039c2cc1397d8a)