asterisk

Commit Graph

Author	SHA1	Message	Date
George Joseph	9bcdef2684	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	5 months ago
George Joseph	b4378ff25e	Prequisites for ARI Outbound Websockets stasis: * Added stasis_app_is_registered(). * Added stasis_app_control_mark_failed(). * Added stasis_app_control_is_failed(). * Fixed res_stasis_device_state so unsubscribe all works properly. * Modified stasis_app_unregister() to unsubscribe from all event sources. * Modified stasis_app_exec to return -1 if stasis_app_control_is_failed() returns true. http: * Added ast_http_create_basic_auth_header(). md5: * Added define for MD5_DIGEST_LENGTH. tcptls: * Added flag to ast_tcptls_session_args to suppress connection log messages to give callers more control over logging. http_websocket: * Add flag to ast_websocket_client_options to suppress connection log messages to give callers more control over logging. * Added username and password to ast_websocket_client_options to support outbound basic authentication. * Added ast_websocket_result_to_str(). (cherry picked from commit `9d75b0447f`)	6 months ago
Peter Jannesen	a3fbcc817b	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 (cherry picked from commit `177b8b6909`)	6 months ago
Joshua C. Colp	725580e79e	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 (cherry picked from commit `8c5c1dcedd`)	6 months ago
George Joseph	7b22d2ef88	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. (cherry picked from commit `abd96ac9c1`)	6 months ago
George Joseph	2a55f064f3	ARI: REST over Websocket This commit adds the ability to make ARI REST requests over the same websocket used to receive events. For full details on how to use the new capability, visit... https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/ Changes: * Added utilities to http.c: * ast_get_http_method_from_string(). * ast_http_parse_post_form(). * Added utilities to json.c: * ast_json_nvp_array_to_ast_variables(). * ast_variables_to_json_nvp_array(). * Added definitions for new events to carry REST responses. * Created res/ari/ari_websocket_requests.c to house the new request handlers. * Moved non-event specific code out of res/ari/resource_events.c into res/ari/ari_websockets.c * Refactored res/res_ari.c to move non-http code out of ast_ari_callback() (which is http specific) and into ast_ari_invoke() so it can be shared between both the http and websocket transports. UpgradeNote: This commit adds the ability to make ARI REST requests over the same websocket used to receive events. See https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/ (cherry picked from commit `1442c17141`)	6 months ago
mkmer	68d9a69be3	audiohook.c: Add ability to adjust volume with float Add the capability to audiohook for float type volume adjustments. This allows for adjustments to volume smaller than 6dB. With INT adjustments, the first step is 2 which converts to ~6dB (or 1/2 volume / double volume depending on adjustment sign). 3dB is a typical adjustment level which can now be accommodated with an adjustment value of 1.41. This is accomplished by the following: Convert internal variables to type float. Always use ast_frame_adjust_volume_float() for adjustments. Cast int to float in original functions ast_audiohook_volume_set(), and ast_volume_adjust(). Cast float to int in ast_audiohook_volume_get() Add functions ast_audiohook_volume_get_float, ast_audiohook_volume_set_float, and ast_audiohook_volume_adjust_float. This update maintains 100% backward compatibility. Resolves: #1171 (cherry picked from commit `ecd1a727e8`)	6 months ago
Allan Nathanson	bdc925c8cc	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 (cherry picked from commit `e2e36748f5`)	6 months ago
Ben Ford	10ef135e5a	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. (cherry picked from commit `091171c0eb`)	7 months ago
Sean Bright	8e900950b6	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. (cherry picked from commit `f042fb2153`)	7 months ago
Allan Nathanson	2d75ac2c0e	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 (cherry picked from commit `ffb0969a98`)	7 months ago
George Joseph	d184ae428d	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 (cherry picked from commit `d4ece751c5`)	7 months ago
Luz Paz	3692e33898	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` (cherry picked from commit `b96640916d`)	7 months ago
George Joseph	de3d6c03e3	bridging: Fix multiple bridging issues causing SEGVs and FRACKs. Issues: * The bridging core allowed multiple bridges to be created with the same unique bridgeId at the same time. Only the last bridge created with the duplicate name was actually saved to the core bridges container. * The bridging core was creating a stasis topic for the bridge and saving it in the bridge->topic field but not increasing its reference count. In the case where two bridges were created with the same uniqueid (which is also the topic name), the second bridge would get the _existing_ topic the first bridge created. When the first bridge was destroyed, it would take the topic with it so when the second bridge attempted to publish a message to it it either FRACKed or SEGVd. * The bridge destructor, which also destroys the bridge topic, is run from the bridge manager thread not the caller's thread. This makes it possible for an ARI developer to create a new one with the same uniqueid believing the old one was destroyed when, in fact, the old one's destructor hadn't completed. This could cause the new bridge to get the old one's topic just before the topic was destroyed. When the new bridge attempted to publish a message on that topic, asterisk could either FRACK or SEGV. * The ARI bridges resource also allowed multiple bridges to be created with the same uniqueid but it kept the duplicate bridges in its app_bridges container. This created a situation where if you added two bridges with the same "bridge1" uniqueid, all operations on "bridge1" were performed on the first bridge created and the second was basically orphaned. If you attempted to delete what you thought was the second bridge, you actually deleted the first one created. Changes: * A new API `ast_bridge_topic_exists(uniqueid)` was created to determine if a topic already exists for a bridge. * `bridge_base_init()` in bridge.c and `ast_ari_bridges_create()` in resource_bridges.c now call `ast_bridge_topic_exists(uniqueid)` to check if a bridge with the requested uniqueid already exists and will fail if it does. * `bridge_register()` in bridges.c now checks the core bridges container to make sure a bridge doesn't already exist with the requested uniqueid. Although most callers of `bridge_register()` will have already called `bridge_base_init()`, which will now fail on duplicate bridges, there is no guarantee of this so we must check again. * The core bridges container allocation was changed to reject duplicate uniqueids instead of silently replacing an existing one. This is a "belt and suspenders" check. * A global mutex was added to bridge.c to prevent concurrent calls to `bridge_base_init()` and `bridge_register()`. * Even though you can no longer create multiple bridges with the same uniqueid at the same time, it's still possible that the bridge topic might be destroyed while a second bridge with the same uniqueid was trying to use it. To address this, the bridging core now increments the reference count on bridge->topic when a bridge is created and decrements it when the bridge is destroyed. * `bridge_create_common()` in res_stasis.c now checks the stasis app_bridges container to make sure a bridge with the requested uniqueid doesn't already exist. This may seem like overkill but there are so many entrypoints to bridge creation that we need to be safe and catch issues as soon in the process as possible. * The stasis app_bridges container allocation was changed to reject duplicate uniqueids instead of adding them. This is a "belt and suspenders" check. * The `bridge show all` CLI command now shows the bridge name as well as the bridge id. * Response code 409 "Conflict" was added as a possible response from the ARI bridge create resources to signal that a bridge with the requested uniqueid already exists. * Additional debugging was added to multiple bridging and stasis files. Resolves: #211 (cherry picked from commit `169b9b9a4f`)	7 months ago
Mike Bradeen	ee09b289ab	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 (cherry picked from commit `7e694417ad`)	7 months ago
Holger Hans Peter Freyther	fe5ff3021e	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. (cherry picked from commit `a0d0c47d06`)	7 months ago
Sean Bright	8f97da052d	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. (cherry picked from commit `46017ea15b`)	7 months ago
Sean Bright	8a3a2958c1	docs: Indent <since> tags. Also updates the 'since' of applications/functions that existed before XML documentation was introduced (1.6.2.0). (cherry picked from commit `67e89b3e77`)	7 months ago
George Joseph	ce550fc1b0	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 (cherry picked from commit `f6a193e87e`)	9 months ago
George Joseph	b794dda76e	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. (cherry picked from commit `936e88512e`)	9 months ago
George Joseph	7a9d014644	README.md, asterisk.c: Update Copyright Dates (cherry picked from commit `d52cd705d3`)	9 months ago
George Joseph	3df8b857ea	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. (cherry picked from commit `772221c82a`)	9 months ago
Allan Nathanson	0fabaa5af4	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 (cherry picked from commit `7e58e8d4f6`)	9 months ago
George Joseph	3a45e487cd	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". (cherry picked from commit `15ed357747`)	9 months ago
Sean Bright	8b736f2047	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 (cherry picked from commit `dafac1f6f4`)	9 months ago
George Joseph	2b428734fd	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. (cherry picked from commit `3c867e6e6c`)	9 months ago
Sean Bright	45f51341db	manager: Add `<since>` tags for all AMI actions. (cherry picked from commit `131682c2c5`)	9 months ago
Steffen Arntz	63c86adbf5	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 (cherry picked from commit `d31b79b2f8`)	9 months ago
Sean Bright	076ec99c23	manager.c: Rename restrictedFile to is_restricted_file. Also correct the spelling of 'privileges.' (cherry picked from commit `e4635e817f`)	9 months ago
Allan Nathanson	139bdb25e4	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 (cherry picked from commit `1cb741df4a`)	9 months ago
Sean Bright	f680ce9d7b	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 (cherry picked from commit `6b3dadfbe2`)	9 months ago
Naveen Albert	3fe5aafd03	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 (cherry picked from commit `afd15274e7`)	9 months ago
George Joseph	bf0a6aaf3b	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. (cherry picked from commit `2fa9dcd51c`)	9 months ago
George Joseph	0a5a6995af	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. (cherry picked from commit `9a1180c6af`)	9 months ago
Tinet-mucw	1a71ea7456	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 (cherry picked from commit `3ac5f82d49`)	9 months ago
Mike Pultz	2d4460ad4d	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. (cherry picked from commit `0c193b725b`)	9 months ago
James Terhune	fc390cf785	main/stasis_channels.c: Fix crash when setting a global variable with invalid UTF8 characters Add check for null value of chan before referencing it with ast_channel_name() Resolves: #999 (cherry picked from commit `d6b7554012`)	9 months ago
Ben Ford	4d80d7ab22	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.	10 months ago
George Joseph	2f8d008103	core_unreal.c: Fix memory leak in ast_unreal_new_channels() When the channel tech is multistream capable, the reference to chan_topology was passed to the new channel. When the channel tech isn't multistream capable, the reference to chan_topology was never released. "Local" channels are multistream capable so it didn't affect them but the confbridge "CBAnn" and the bridge_media "Recorder" channels are not so they caused a leak every time one of them was created. Also added tracing to ast_stream_topology_alloc() and stream_topology_destroy() to assist with debugging. Resolves: #938 (cherry picked from commit `41162f963b`)	12 months ago
Allan Nathanson	039baa0317	dnsmgr.c: dnsmgr_refresh() incorrectly flags change with DNS round-robin The dnsmgr_refresh() function checks to see if the IP address associated with a name/service has changed. The gotcha is that the ast_get_ip_or_srv() function only returns the first IP address returned by the DNS query. If there are multiple IPs associated with the name and the returned order is not consistent (e.g. with DNS round-robin) then the other IP addresses are not included in the comparison and the entry is flagged as changed even though the IP is still valid. Updated the code to check all IP addresses and flag a change only if the original IP is no longer valid. Resolves: #924 (cherry picked from commit `338e191539`)	12 months ago
George Joseph	04b882397f	manager.c: Add unit test for Originate app and appdata permissions This unit test checks that dialplan apps and app data specified as parameters for the Originate action are allowed with the permissions the user has. (cherry picked from commit `9ecccb21ed`)	12 months ago
Sean Bright	682ad186fb	res_agi.c: Ensure SIGCHLD handler functions are properly balanced. Calls to `ast_replace_sigchld()` and `ast_unreplace_sigchld()` must be balanced to ensure that we can capture the exit status of child processes when we need to. This extends to functions that call `ast_replace_sigchld()` and `ast_unreplace_sigchld()` such as `ast_safe_fork()` and `ast_safe_fork_cleanup()`. The primary change here is ensuring that we do not call `ast_safe_fork_cleanup()` in `res_agi.c` if we have not previously called `ast_safe_fork()`. Additionally we reinforce some of the documentation and add an assertion to, ideally, catch this sooner were this to happen again. Fixes #922 (cherry picked from commit `e1f2866bb6`)	12 months ago
Naveen Albert	ab79321a29	main, res, tests: Fix compilation errors on FreeBSD. asterisk.c, manager.c: Increase buffer sizes to avoid truncation warnings. config.c: Include header file for WIFEXITED/WEXITSTATUS macros. res_timing_kqueue: Use more portable format specifier. test_crypto: Use non-linux limits.h header file. Resolves: #916 (cherry picked from commit `b939d840db`)	12 months ago
Ben Ford	c7beb10006	manager.c: Restrict ModuleLoad to the configured modules directory. When using the ModuleLoad AMI action, it was possible to traverse upwards through the directories to files outside of the configured modules directory. We decided it would be best to restrict access to modules exclusively in the configured directory. You will now get an error when the specified module is outside of this limitation. Fixes: #897 UserNote: The ModuleLoad AMI action now restricts modules to the configured modules directory. (cherry picked from commit `ee9a0f056e`)	12 months ago
Naveen Albert	1372eeba51	astfd.c: Avoid calling fclose with NULL argument. Don't pass through a NULL argument to fclose, which is undefined behavior, and instead return -1 and set errno appropriately. This also avoids a compiler warning with glibc 2.38 and newer, as glibc commit 71d9e0fe766a3c22a730995b9d024960970670af added the nonnull attribute to this argument. Resolves: #900 (cherry picked from commit `2c583be963`)	12 months ago
Peter Jannesen	2f44ac22b2	channel: Preserve CHANNEL(userfield) on masquerade. In certain circumstances a channel may undergo an operation referred to as a masquerade. If this occurs the CHANNEL(userfield) value was not preserved causing it to get lost. This change makes it so that this field is now preserved. Fixes: #882 (cherry picked from commit `fd196e9113`)	12 months ago
George Joseph	6dc00f642a	Fix application references to Background The app is actually named "BackGround" but several references in XML documentation were spelled "Background" with the lower case "g". This was causing documentation links to return "not found" messages. (cherry picked from commit `f71668f0b9`)	12 months ago
George Joseph	56d3070939	manager: Enhance event filtering for performance UserNote: You can now perform more granular filtering on events in manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with)) = PJSIP/` This is much more efficient than `eventfilter = Event: Newchannel.*Channel: PJSIP/` Full syntax guide is in configs/samples/manager.conf.sample. (cherry picked from commit `b8d16fd800`)	12 months ago
George Joseph	2ca93f0cdc	manager.c: Split XML documentation to manager_doc.xml (cherry picked from commit `dd643bf827`)	12 months ago
George Joseph	6297462fd2	db.c: Remove limit on family/key length Consumers like media_cache have been running into issues with the previous astdb "/family/key" limit of 253 bytes when needing to store things like long URIs. An Amazon S3 URI is a good example of this. Now, instead of using a static 256 byte buffer for "/family/key", we use ast_asprintf() to dynamically create it. Both test_db.c and test_media_cache.c were also updated to use keys/URIs over the old 253 character limit. Resolves: #881 UserNote: The `ast_db_()` APIs have had the 253 byte limit on "/family/key" removed and will now accept families and keys with a total length of up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB` dialplan applications, dialplan functions, manager actions and `databse` CLI commands. Since the media_cache also uses the `ast_db_*()` APIs, you can now store resources with URIs longer than 253 bytes. (cherry picked from commit `ff7d094803`)	12 months ago

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6908 Commits (22.4.1)