asterisk

Commit Graph

Author	SHA1	Message	Date
Mike Bradeen	579cd0b8bb	chan_pjsip: Serialize INVITE creation on DTMF attended transfer When a call is transfered via DTMF feature code, the Transfer Target and Transferer are bridged immediately. This opens the possibilty of a race condition between the creation of an INVITE and the bridge induced colp update that can result in the set caller ID being over-written with the transferer's default info. Fixes: #1234	5 months 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	5 months 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.	5 months ago
Naveen Albert	89f7d5a471	app_sms: Ignore false positive vectorization warning. Ignore gcc warning about writing 32 bytes into a region of size 6, since we check that we don't go out of bounds for each byte. This is due to a vectorization bug in gcc 15, stemming from gcc commit 68326d5d1a593dc0bf098c03aac25916168bc5a9. Resolves: #1088	5 months ago
George Joseph	6fa0e264df	lock.h: Add include for string.h when DEBUG_THREADS is defined. When DEBUG_THREADS is defined, lock.h uses strerror(), which is defined in the libc string.h file, to print warning messages. If the including source file doesn't include string.h then strerror() won't be found and and compile errors will be thrown. Since lock.h depends on this, string.h is now included from there if DEBUG_THREADS is defined. This way, including source files don't have to worry about it.	5 months ago
Naveen Albert	754dea319e	res_pjsip_caller_id: Also parse URI parameters for ANI2. If the isup-oli was sent as a URI parameter, rather than a header parameter, it was not being parsed. Make sure we parse both if needed so the ANI2 is set regardless of which type of parameter the isup-oli is sent as. Resolves: #1220	5 months ago
Naveen Albert	923e6d471a	func_callerid: Always format ANI2 as two digits. ANI II is always supposed to be formatted as two digits, so zero pad when formatting it if necessary. Resolves: #1222	5 months ago
Naveen Albert	f6ceeff523	app_meetme: Remove inaccurate removal version from xmldocs. app_meetme is deprecated but wasn't removed as planned in 21, so remove the inaccurate removal version. Resolves: #1224	5 months ago
Luz Paz	9ca3233aea	docs: Fix typos in apps/ Found via codespell	5 months ago
Mike Bradeen	8924258639	stasis/control.c: Set Hangup Cause to No Answer on Dial timeout Other Dial operations (dial, app_dial) use Q.850 cause 19 when a dial timeout occurs, but the Dial command via ARI did not set an explicit reason. This resulted in a CANCEL with Normal Call Clearing and corresponding ChannelDestroyed. This change sets the hangup cause to AST_CAUSE_NO_ANSWER to be consistent with the other operations. Fixes: #963 UserNote: A Dial timeout on POST /channels/{channelId}/dial will now result in a CANCEL and ChannelDestroyed with cause 19 / User alerting, no answer. Previously no explicit cause was set, resulting in a cause of 16 / Normal Call Clearing.	6 months ago
Naveen Albert	d60bcc56a7	chan_iax2: Minor improvements to documentation and warning messages. * Update Dial() documentation for IAX2 to include syntax for RSA public key names. * Add additional details to a couple warnings to provide more context when an undecodable frame is received. Resolves: #1206	6 months ago
Andreas Wehrmann	e5c05d4225	pbx_ael: unregister AELSub application and CLI commands on module load failure This fixes crashes/hangs I noticed with Asterisk 20.3.0 and 20.13.0 and quickly found out, that the AEL module doesn't do proper cleanup when it fails to load. This happens for example when there are syntax errors and AEL fails to compile in which case pbx_load_module() returns an error but load_module() doesn't then unregister CLI cmds and the application.	6 months ago
Albrecht Oster	a9cd7f9b8d	res_pjproject: Fix DTLS client check failing on some platforms Certain platforms (mainly BSD derivatives) have an additional length field in `sockaddr_in6` and `sockaddr_in`. `ast_sockaddr_from_pj_sockaddr()` does not take this field into account when copying over values from the `pj_sockaddr` into the `ast_sockaddr`. The resulting `ast_sockaddr` will have an uninitialized value for `sin6_len`/`sin_len` while the other `ast_sockaddr` (not converted from a `pj_sockaddr`) to check against in `ast_sockaddr_pj_sockaddr_cmp()` has the correct length value set. This has the effect that `ast_sockaddr_cmp()` will always indicate an address mismatch, because it does a bitwise comparison, and all DTLS packets are dropped even if addresses and ports match. `ast_sockaddr_from_pj_sockaddr()` now checks whether the length fields are available on the current platform and sets the values accordingly. Resolves: #505	6 months 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().	6 months ago
Ben Ford	576f6bec3d	contrib: Add systemd service and timer files for malloc trim. Adds two files to the contrib/systemd/ directory that can be installed to periodically run "malloc trim" on Asterisk. These files do nothing unless they are explicitly moved to the correct location on the system. Users who are experiencing Asterisk memory issues can use this service to potentially help combat the problem. These files can also be configured to change the start time and interval. See systemd.timer(5) and systemd.time(7) for more information. UserNote: Service and timer files for systemd have been added to the contrib/systemd/ directory. If you are experiencing memory issues, install these files to have "malloc trim" periodically run on the system.	6 months 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	6 months 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	6 months ago
phoneben	ac07cbe2c3	Add log-caller-id-name option to log Caller ID Name in queue log Add log-caller-id-name option to log Caller ID Name in queue log This patch introduces a new global configuration option, log-caller-id-name, to queues.conf to control whether the Caller ID name is logged when a call enters a queue. When log-caller-id-name=yes, the Caller ID name is logged as parameter 4 in the queue log, provided it’s allowed by the existing log_restricted_caller_id rules. If log-caller-id-name=no (the default), the Caller ID name is omitted from the logs. Fixes: #1091 UserNote: This patch adds a global configuration option, log-caller-id-name, to queues.conf to control whether the Caller ID name is logged as parameter 4 when a call enters a queue. When log-caller-id-name=yes, the Caller ID name is included in the queue log, Any '\|' characters in the caller ID name will be replaced with '_'. (provided it’s allowed by the existing log_restricted_caller_id rules). When log-caller-id-name=no (the default), the Caller ID name is omitted.	6 months 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.	6 months ago
Sean Bright	085fd922fc	app_confbridge: Prevent crash when publishing channel-less event. Resolves: #1190	6 months ago
Sean Bright	9fdf8597cb	channel: Deprecate `ast_moh_cleanup(...)`. We don't want anyone calling it but the channel destructor.	6 months ago
George Joseph	4620e76012	ari_websockets: Fix frack if ARI config fails to load. ari_ws_session_registry_dtor() wasn't checking that the container was valid before running ao2_callback on it to shutdown registered sessions.	6 months 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/	6 months 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	6 months ago
Florent CHAUVEAU	a5bc39fa32	audiosocket: added support for DTMF frames Updated the AudioSocket protocol to allow sending DTMF frames. AST_FRAME_DTMF frames are now forwarded to the server, in addition to AST_FRAME_AUDIO frames. A new payload type AST_AUDIOSOCKET_KIND_DTMF with value 0x03 was added to the protocol. The payload is a 1-byte ascii representing the DTMF digit (0-9,,#...). UserNote: The AudioSocket protocol now forwards DTMF frames with payload type 0x03. The payload is a 1-byte ascii representing the DTMF digit (0-9,,#...).	6 months ago
Norm Harrison	6ef4a2d008	asterisk/channel.h: fix documentation for 'ast_waitfor_nandfds()' Co-authored-by: Florent CHAUVEAU <florentch@pm.me>	6 months ago
Norm Harrison	4e546c35c7	audiosocket: fix timeout, fix dialplan app exit, server address in logs - Correct wait timeout logic in the dialplan application. - Include server address in log messages for better traceability. - Allow dialplan app to exit gracefully on hangup messages and socket closure. - Optimize I/O by reducing redundant read()/write() operations. Co-authored-by: Florent CHAUVEAU <florentch@pm.me>	6 months ago
Mark Murawski	de88e12cb6	chan_pjsip: Add the same details as PJSIPShowContacts to the CLI via 'pjsip show contact' CLI 'pjsip show contact' does not show enough information. One must telnet to AMI or write a script to ask Asterisk for example what the User-Agent is on a Contact This feature adds the same details as PJSIPShowContacts to the CLI Resolves: #643	6 months ago
Zhai Liangliang	8fddb09bec	Update config.guess and config.sub	6 months ago
Alexei Gradinari	1f42d20d9c	chan_pjsip: set correct Endpoint Device State on multiple channels 1. When one channel is placed on hold, the device state is set to ONHOLD without checking other channels states. In case of AST_CONTROL_HOLD set the device state as AST_DEVICE_UNKNOWN to calculate aggregate device state of all active channels. 2. The current implementation incorrectly classifies channels in use. The only channels that has the states: UP, RING and BUSY are considered as "in use". A channel should be considered "in use" if its state is anything other than DOWN or RESERVED. 3. Currently, if the number of channels "in use" is greater than device_state_busy_at, the system does not set the state to BUSY. Instead, it incorrectly assigns an aggregate device state. The endpoint device state should be BUSY if the number of channels "in use" is greater than or equal to device_state_busy_at. Fixes: #1181	6 months 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	6 months 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.	7 months ago
Sean Bright	f918a6fb53	res_config_curl.c: Remove unnecessary warnings. Resolves: #1164	7 months ago
George Joseph	606d5d054a	README.md: Updates and Fixes * Outdated information has been removed. * New links added. * Placeholder added for link to change logs. Going forward, the release process will create HTML versions of the README and change log and will update the link in the README to the current change log for the branch... * In the development branches, the link will always point to the current release on GitHub. * In the "releases/" branches and the tarballs, the link will point to the ChangeLogs/ChangeLog-<version>.html file in the source directory. On the downloads website, the link will point to the ChangeLog-<version>.html file in the same directory. Resolves: #1131	7 months ago
Sean Bright	777cd31d22	res_rtp_asterisk.c: Don't truncate spec-compliant `ice-ufrag` or `ice-pwd`. RFC 8839[1] indicates that the `ice-ufrag` and `ice-pwd` attributes can be up to 256 bytes long. While we don't generate values of that size, we should be able to accomodate them without truncating. 1. https://www.rfc-editor.org/rfc/rfc8839#name-ice-ufrag-and-ice-pwd-attri	7 months ago
Joshua Elson	2f6578925d	fix: Correct default flag for tcp_keepalive_enable option Resolves an issue where the tcp_keepalive_enable option was not properly enabled in the sample configuration due to an incorrect default flag setting. Fixes: #1149	7 months 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	7 months 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.	7 months 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	7 months 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	7 months ago
George Joseph	5470a23b48	swagger_model.py: Fix invalid escape sequence in get_list_parameter_type(). Recent python versions complain when backslashes in strings create invalid escape sequences. This causes issues for strings used as regex patterns like `'^List\[(.)\]$'` where you want the regex parser to treat `[` and `]` as literals. Double-backslashing is one way to fix it but simply converting the string to a raw string `re.match(r'^List\[(.)\]$', text)` is easier and less error prone.	7 months ago
Maximilian Fridrich	e02de88e4e	Revert "res_rtp_asterisk.c: Set Mark on rtp when timestamp skew is too big" This reverts commit `f30ad96b3f`. The original change was not RFC compliant and caused issues because it set the RTP marker bit in cases when it shouldn't be set. See the linked issue #1135 for a detailed explanation. Fixes: #1135.	7 months ago
Sean Bright	9665903d4b	res_rtp_asterisk.c: Use correct timeout value for T.140 RED timer. Found while reviewing #1128	7 months ago
Luz Paz	2d7948fa48	docs: Fix typos in cdr/ Found via codespell	8 months ago
Luz Paz	d65af637ac	docs: Fix various typos in channels/ Found via `codespell -q 3 -S "./CREDITS,*.po" -L abd,asent,atleast,cachable,childrens,contentn,crypted,dne,durationm,enew,exten,inout,leapyear,mye,nd,oclock,offsetp,ot,parm,parms,preceeding,pris,ptd,requestor,re-use,re-used,re-uses,ser,siz,slanguage,slin,thirdparty,varn,varns,ues`	8 months 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`	8 months 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	8 months ago
George Joseph	4075d35ed3	.github: Change concurrency group ids so they're unique. GitHub strikes again. Apparently the github.ref context variable only contains the PR number if the workflow is triggered by "pull_request" so since we just changed the trigger to "pull_request_target" the variable no longer contains the PR number and is therefore not unique and can't be used as a concurrency group id. We now use `github.triggering_actor-github.head_ref`.	8 months 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	8 months ago
George Joseph	461e9ac304	.github: Refactor Releaser to use reusable workflow	8 months ago

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