6922 Commits (d5bd2b3ce952e7e7e49a2b787ca16fe8ebe95bfe)

Author	SHA1	Message	Date
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
Luz Paz	e3ad4cea4e	docs: Fix various typos in main/ ... Found via `codespell -q 3 -S "./CREDITS" -L abd,asent,atleast,childrens,contentn,crypted,dne,durationm,exten,inout,leapyear,nd,oclock,offsetp,ot,parm,parms,requestor,ser,slanguage,slin,thirdparty,varn,varns,ues`	2 years ago
George Joseph	076799be15	bridging: Fix multiple bridging issues causing SEGVs and FRACKs. ... Issues: * The bridging core allowed multiple bridges to be created with the same unique bridgeId at the same time. Only the last bridge created with the duplicate name was actually saved to the core bridges container. * The bridging core was creating a stasis topic for the bridge and saving it in the bridge->topic field but not increasing its reference count. In the case where two bridges were created with the same uniqueid (which is also the topic name), the second bridge would get the _existing_ topic the first bridge created. When the first bridge was destroyed, it would take the topic with it so when the second bridge attempted to publish a message to it it either FRACKed or SEGVd. * The bridge destructor, which also destroys the bridge topic, is run from the bridge manager thread not the caller's thread. This makes it possible for an ARI developer to create a new one with the same uniqueid believing the old one was destroyed when, in fact, the old one's destructor hadn't completed. This could cause the new bridge to get the old one's topic just before the topic was destroyed. When the new bridge attempted to publish a message on that topic, asterisk could either FRACK or SEGV. * The ARI bridges resource also allowed multiple bridges to be created with the same uniqueid but it kept the duplicate bridges in its app_bridges container. This created a situation where if you added two bridges with the same "bridge1" uniqueid, all operations on "bridge1" were performed on the first bridge created and the second was basically orphaned. If you attempted to delete what you thought was the second bridge, you actually deleted the first one created. Changes: * A new API `ast_bridge_topic_exists(uniqueid)` was created to determine if a topic already exists for a bridge. * `bridge_base_init()` in bridge.c and `ast_ari_bridges_create()` in resource_bridges.c now call `ast_bridge_topic_exists(uniqueid)` to check if a bridge with the requested uniqueid already exists and will fail if it does. * `bridge_register()` in bridges.c now checks the core bridges container to make sure a bridge doesn't already exist with the requested uniqueid. Although most callers of `bridge_register()` will have already called `bridge_base_init()`, which will now fail on duplicate bridges, there is no guarantee of this so we must check again. * The core bridges container allocation was changed to reject duplicate uniqueids instead of silently replacing an existing one. This is a "belt and suspenders" check. * A global mutex was added to bridge.c to prevent concurrent calls to `bridge_base_init()` and `bridge_register()`. * Even though you can no longer create multiple bridges with the same uniqueid at the same time, it's still possible that the bridge topic might be destroyed while a second bridge with the same uniqueid was trying to use it. To address this, the bridging core now increments the reference count on bridge->topic when a bridge is created and decrements it when the bridge is destroyed. * `bridge_create_common()` in res_stasis.c now checks the stasis app_bridges container to make sure a bridge with the requested uniqueid doesn't already exist. This may seem like overkill but there are so many entrypoints to bridge creation that we need to be safe and catch issues as soon in the process as possible. * The stasis app_bridges container allocation was changed to reject duplicate uniqueids instead of adding them. This is a "belt and suspenders" check. * The `bridge show all` CLI command now shows the bridge name as well as the bridge id. * Response code 409 "Conflict" was added as a possible response from the ARI bridge create resources to signal that a bridge with the requested uniqueid already exists. * Additional debugging was added to multiple bridging and stasis files. Resolves: #211	2 years ago
Mike Bradeen	d63c3e80fc	bridge_channel: don't set cause code on channel during bridge delete if already set ... Due to a potential race condition via ARI when hanging up a channel hangup with cause while also deleting a bridge containing that channel, the bridge delete can over-write the hangup cause code resulting in Normal Call Clearing instead of the set value. With this change, bridge deletion will only set the hangup code if it hasn't been previously set. Resolves: #1124	2 years ago
Holger Hans Peter Freyther	9fc631d028	ari/pjsip: Make it possible to control transfers through ARI ... Introduce a ChannelTransfer event and the ability to notify progress to ARI. Implement emitting this event from the PJSIP channel instead of handling the transfer in Asterisk when configured. Introduce a dialplan function to the PJSIP channel to switch between the "core" and "ari-only" behavior. UserNote: Call transfers on the PJSIP channel can now be controlled by ARI. This can be enabled by using the PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.	2 years ago
Sean Bright	e200f51ea4	channel.c: Remove dead AST_GENERATOR_FD code. ... Nothing ever sets the `AST_GENERATOR_FD`, so this block of code will never execute. It also is the only place where the `generate` callback is called with the channel lock held which made it difficult to reason about the thread safety of `ast_generator`s. In passing, also note that `AST_AGENT_FD` isn't used either.	2 years ago
Sean Bright	03f1c24674	utils: Remove libdb and astdb conversion scripts. ... These were included with Asterisk 10 when we switched astdb from libdb to sqlite3.	2 years ago
Sean Bright	82bef2ed90	docs: Indent <since> tags. ... Also updates the 'since' of applications/functions that existed before XML documentation was introduced (1.6.2.0).	2 years ago
George Joseph	198b01c536	docs: Add version information to application and function XML elements ... * Do a git blame on the embedded XML application or function element. * From the commit hash, grab the summary line. * Do a git log --grep <summary> to find the cherry-pick commits in all branches that match. * Do a git patch-id to ensure the commits are all related and didn't get a false match on the summary. * Do a git tag --contains <commit> to find the tags that contain each commit. * Weed out all tags not ..0. * Sort and discard any .0.0 and following tags where the commit appeared in an earlier branch. * The result is a single tag for each branch where the application or function was defined. The applications and functions defined in the following files were done by hand because the XML was extracted from the C source file relatively recently. * channels/pjsip/dialplan_functions_doc.xml * main/logger_doc.xml * main/manager_doc.xml * res/res_geolocation/geoloc_doc.xml * res/res_stir_shaken/stir_shaken_doc.xml	2 years ago
George Joseph	77d5550d14	docs: Add version information to manager event instance XML elements ... * Do a git blame on the embedded XML managerEvent elements. * From the commit hash, grab the summary line. * Do a git log --grep <summary> to find the cherry-pick commits in all branches that match. * Do a git patch-id to ensure the commits are all related and didn't get a false match on the summary. * Do a git tag --contains <commit> to find the tags that contain each commit. * Weed out all tags not ..0. * Sort and discard any .0.0 and following tags where the commit appeared in an earlier branch. * The result is a single tag for each branch where the application or function was defined. The events defined in res/res_pjsip/pjsip_manager.xml were done by hand because the XML was extracted from the C source file relatively recently. Two bugs were fixed along the way... * The get_documentation awk script was exiting after it processed the first DOCUMENTATION block it found in a file. We have at least 1 source file with multiple DOCUMENTATION blocks so only the first one in them was being processed. The awk script was changed to continue searching rather than exiting after the first block. * Fixing the awk script revealed an issue in logger.c where the third DOCUMENTATION block contained a XML fragment that consisted only of a managerEventInstance element that wasn't wrapped in a managerEvent element. Since logger_doc.xml already existed, the remaining fragments in logger.c were moved to it and properly organized.	2 years ago
George Joseph	6384e8667c	README.md, asterisk.c: Update Copyright Dates	2 years ago
George Joseph	316693ded9	docs: Add version information to configObject and configOption XML elements ... Most of the configObjects and configOptions that are implemented with ACO or Sorcery now have `<since>/<version>` elements added. There are probably some that the script I used didn't catch. The version tags were determined by the following... * Do a git blame on the API call that created the object or option. * From the commit hash, grab the summary line. * Do a `git log --grep <summary>` to find the cherry-pick commits in all branches that match. * Do a `git patch-id` to ensure the commits are all related and didn't get a false match on the summary. * Do a `git tag --contains <commit>` to find the tags that contain each commit. * Weed out all tags not <major>.<minor>.0. * Sort and discard any <major>.0.0 and following tags where the commit appeared in an earlier branch. * The result is a single tag for each branch where the API was last touched. configObjects and configOptions elements implemented with the base ast_config APIs were just not possible to find due to the non-deterministic way they are accessed. Also note that if the API call was on modified after it was added, the version will be the one it was last modified in. Final note: The configObject and configOption elements were introduced in 12.0.0 so options created before then may not have any XML documentation.	2 years ago
Allan Nathanson	217e696288	config.c: fix #tryinclude being converted to #include on rewrite ... Correct an issue in ast_config_text_file_save2() when updating configuration files with "#tryinclude" statements. The API currently replaces "#tryinclude" with "#include". The API also creates empty template files if the referenced files do not exist. This change resolves these problems. Resolves: https://github.com/asterisk/asterisk/issues/920	2 years ago
George Joseph	114fa12f05	docs: Various XML fixes ... * channels/pjsip/dialplan_functions_doc.xml: Added xmlns:xi to docs element. * main/bucket.c: Removed XML completely since the "bucket" and "file" objects are internal only with no config file. * main/named_acl.c: Fixed the configFile element name. It was "named_acl.conf" and should have been "acl.conf" * res/res_geolocation/geoloc_doc.xml: Added xmlns:xi to docs element. * res/res_http_media_cache.c: Fixed the configFile element name. It was "http_media_cache.conf" and should have been "res_http_media_cache.conf".	2 years ago
Sean Bright	1047291662	strings.c: Improve numeric detection in `ast_strings_match()`. ... Essentially, we were treating 1234x1234 and 1234x5678 as 'equal' because we were able to convert the prefix of each of these strings to the same number. Resolves: #1028	2 years ago
George Joseph	3868f7cb47	docs: Enable since/version handling for XML, CLI and ARI documentation ... * Added the "since" element to the XML configObject and configOption elements in appdocsxml.dtd. * Added the "Since" section to the following CLI output: ``` config show help <module> <object> config show help <module> <object> <option> core show application <app> core show function <func> manager show command <command> manager show event <event> agi show commands topic <topic> ``` * Refactored the commands above to output their sections in the same order: Synopsis, Since, Description, Syntax, Arguments, SeeAlso * Refactored the commands above so they all use the same pattern for writing the output to the CLI. * Fixed several memory leaks caused by failure to free temporary output buffers. * Added a "since" array to the mustache template for the top-level resources (Channel, Endpoint, etc.) and to the paths/methods underneath them. These will be added to the generated markdown if present. Example: ``` "resourcePath": "/api-docs/channels.{format}", "requiresModules": [ "res_stasis_answer", "res_stasis_playback", "res_stasis_recording", "res_stasis_snoop" ], "since": [ "18.0.0", "21.0.0" ], "apis": [ { "path": "/channels", "description": "Active channels", "operations": [ { "httpMethod": "GET", "since": [ "18.6.0", "21.8.0" ], "summary": "List all active channels in Asterisk.", "nickname": "list", "responseClass": "List[Channel]" }, ``` NOTE: No versioning information is actually added in this commit. Those will be added separately and instructions for adding and maintaining them will be published on the documentation site at a later date.	2 years ago
Sean Bright	468601fd13	manager: Add `<since>` tags for all AMI actions.	2 years ago
Steffen Arntz	4ef68fbce8	logger.c fix: malformed JSON template ... this typo was mentioned before, but never got fixed. https://community.asterisk.org/t/logger-cannot-log-long-json-lines-properly/87618/6	2 years ago
Sean Bright	7b71efd172	manager.c: Rename restrictedFile to is_restricted_file. ... Also correct the spelling of 'privileges.'	2 years ago
Allan Nathanson	22b261575f	config.c: retain leading whitespace before comments ... Configurations loaded with the ast_config_load2() API and later written out with ast_config_text_file_save2() will have any leading whitespace stripped away. The APIs should make reasonable efforts to maintain the content and formatting of the configuration files. This change retains any leading whitespace from comment lines that start with a ";". Resolves: https://github.com/asterisk/asterisk/issues/970	2 years ago
Ben Ford	7f70190c59	manager.c: Restrict ListCategories to the configuration directory. ... When using the ListCategories AMI action, it was possible to traverse upwards through the directories to files outside of the configured configuration directory. This action is now restricted to the configured directory and an error will now be returned if the specified file is outside of this limitation. Resolves: #GHSA-33x6-fj46-6rfh UserNote: The ListCategories AMI action now restricts files to the configured configuration directory.	2 years ago
Sean Bright	e904ae5df0	config.c: Fix off-nominal reference leak. ... This was identified and fixed by @Allan-N in #918 but it is an important fix in its own right. The fix here is slightly different than Allan's in that we just move the initialization of the problematic AO2 container to where it is first used. Fixes #1046	2 years ago
Naveen Albert	5ed7f4d152	chan_dahdi: Fix wrong channel state when RINGING recieved. ... Previously, when AST_CONTROL_RINGING was received by a DAHDI device, it would set its channel state to AST_STATE_RINGING. However, an analysis of the codebase and other channel drivers reveals RINGING corresponds to physical power ringing, whereas AST_STATE_RING should be used for audible ringback on the channel. This also ensures the correct device state is returned by the channel state to device state conversion. Since there seems to be confusion in various places regarding AST_STATE_RING vs. AST_STATE_RINGING, some documentation has been added or corrected to clarify the actual purposes of these two channel states, and the associated device state mapping. An edge case that prompted this fix, but isn't explicitly addressed here, is that of an incoming call to an FXO port. The channel state will be "Ring", which maps to a device state of "In Use", not "Ringing" as would be more intuitive. However, this is semantic, since technically, Asterisk is treating this the same as any other incoming call, and so "Ring" is the semantic state (put another way, Asterisk isn't ringing anything, like in the cases where channels are in the "Ringing" state). Since FXO ports don't currently support Call Waiting, a suitable workaround for the above would be to ignore the device state and instead check the channel state (e.g. IMPORT(DAHDI/1-1,CHANNEL(state))) since it will be Ring if the FXO port is idle (but a call is ringing on it) and Up if the FXO port is actually in use. (In both cases, the device state would misleadingly be "In Use".) Resolves: #1029	2 years ago
George Joseph	92e5ed5a3a	Add ability to pass arguments to unit tests from the CLI ... Unit tests can now be passed custom arguments from the command line. For example, the following command would run the "mytest" test in the "/main/mycat" category with the option "myoption=54" `CLI> test execute category /main/mycat name mytest options myoption=54` You can also pass options to an entire category... `CLI> test execute category /main/mycat options myoption=54` Basically, everything after the "options" keyword is passed verbatim to the test which must decide what to do with it. * A new API ast_test_get_cli_args() was created to give the tests access to the cli_args->argc and cli_args->argv elements. * Although not needed for the option processing, a new macro ast_test_validate_cleanup_custom() was added to test.h that allows you to specify a custom error message instead of just "Condition failed". * The test_skel.c was updated to demonstrate parsing options and the use of the ast_test_validate_cleanup_custom() macro.	2 years ago
George Joseph	92f361c826	Allow C++ source files (as extension .cc) in the main directory ... Although C++ files (as extension .cc) have been handled in the module directories for many years, the main directory was missing one line in its Makefile that prevented C++ files from being recognised there.	2 years ago
Naveen Albert	c157ee62f6	config.c: Make ast_variable_retrieve return last match. ... ast_variable_retrieve currently returns the first match for a variable, as opposed to the last one. This is problematic because modules that load config settings by explicitly calling ast_variable_retrieve on a variable name (as opposed to iterating through all the directives as specified) will end up taking the first specified value, such as the default value from the template rather than the actual effective value in an individual config section, leading to the wrong config. This fixes this by making ast_variable_retrieve return the last match, or the most recently overridden one, as the effective setting. This is similar to what the -1 index in the AST_CONFIG function does. There is another function, ast_variable_find_last_in_list, that does something similar. However, it's a slightly different API, and it sees virtually no usage in Asterisk. ast_variable_retrieve is what most things use so this is currently the relevant point of breakage. In practice, this is unlikely to cause any breakage, since there would be no logical reason to use an inherited value rather than an explicitly overridden value when loading a config. ASTERISK-30370 #close Resolves: #244 UpgradeNote: Config variables retrieved explicitly by name now return the most recently overriding value as opposed to the base value (e.g. from a template). This is equivalent to retrieving a config setting using the -1 index to the AST_CONFIG function. The major implication of this is that modules processing configs by explicitly retrieving variables by name will now get the effective value of a variable as overridden in a config rather than the first-set value (from a template), which is consistent with how other modules load config settings.	2 years ago
Tinet-mucw	a23f28e7e0	audiohook.c: resolving the issue with audiohook both reading when packet loss on one side of the call ... When there is 0% packet loss on one side of the call and 15% packet loss on the other side, reading frame is often failed when reading direction_both audiohook. when read_factory available = 0, write_factory available = 320; i think write factory is usable read; because after reading one frame, there is still another frame that can be read together with the next read factory frame. Resolves: #851	2 years ago
Mike Pultz	e020ee5795	manager.c: Add Processed Call Count to CoreStatus output ... This update adds the processed call count to the CoreStatus AMI Action responsie. This output is similar to the values returned by "core show channels" or "core show calls" in the CLI. UserNote: The current processed call count is now returned as CoreProcessedCalls from the CoreStatus AMI Action.	2 years ago