asterisk

Commit Graph

Author	SHA1	Message	Date
Sean Bright	3289aa79db	Merge `40ea6a0e30` into `b4edd3ab7d`	2 months ago
Luz Paz	40ea6a0e30	docs: Fix typos in apps/ Found via codespell	2 months ago
Mike Bradeen	b4edd3ab7d	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.	2 months ago
Naveen Albert	7724e5e6ad	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	2 months ago
Andreas Wehrmann	aca78d4936	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.	2 months ago
Albrecht Oster	5f9cd260ac	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	2 months ago
George Joseph	cc92adc5fb	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().	2 months ago
Ben Ford	6925b0118e	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.	2 months ago
Peter Jannesen	f9e5b9aa01	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	2 months ago
Joshua C. Colp	ca341910e7	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	2 months ago
phoneben	f843f24be4	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.	2 months ago
George Joseph	aacabb48ee	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.	2 months ago
Sean Bright	efeb102507	app_confbridge: Prevent crash when publishing channel-less event. Resolves: #1190	2 months ago
George Joseph	234ca964e5	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.	2 months ago
George Joseph	4e07d3d50f	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/	2 months ago
mkmer	874fde5924	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	2 months ago
Florent CHAUVEAU	c44e7e85ec	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,,#...).	3 months ago
Norm Harrison	5cfc79ebea	asterisk/channel.h: fix documentation for 'ast_waitfor_nandfds()' Co-authored-by: Florent CHAUVEAU <florentch@pm.me>	3 months ago
Norm Harrison	218c64c4eb	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>	3 months ago
Mark Murawski	51f9e1f1ba	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	3 months ago
Zhai Liangliang	c9a85f7124	Update config.guess and config.sub	3 months ago
Alexei Gradinari	ad178d155d	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	3 months ago
Allan Nathanson	df8c32f092	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	3 months ago
Ben Ford	11684bc0d3	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.	3 months ago
Sean Bright	92b005abf8	res_config_curl.c: Remove unnecessary warnings. Resolves: #1164	3 months ago
George Joseph	92e02c1574	res_pjsip_outbound_registration.c: Remove xpointer reference to user_agent. Commit `e86f937e` added AMI documentation to the module but the cherry-pick from master to the 20 branch pulled in an xpointer reference to the `user_agent` config option which doesn't actually exist in 20. This causes asterisk to fail to start because it can't load the core-en_US.xml documentation file. The CI cherry-pick tests should have caught this but there was bug in the cherry-picker that was causing the commits to be cherry-picked to the wrong branch. Just removing the xpointer reference resolves the issue.	3 months ago
George Joseph	9f4b52bf2b	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	3 months ago
Sean Bright	3cafe10d21	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	3 months ago
Joshua Elson	53f3ac7925	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	3 months ago
Sean Bright	e86f937e3a	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.	3 months ago
Allan Nathanson	eec9bed8af	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	3 months ago
George Joseph	266440ab20	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	3 months ago
George Joseph	cc267af3e1	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.	3 months ago
Maximilian Fridrich	5565d00eb4	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.	3 months ago
Sean Bright	ccc596ab5f	res_rtp_asterisk.c: Use correct timeout value for T.140 RED timer. Found while reviewing #1128	3 months ago
Luz Paz	43599cfb2a	docs: Fix typos in cdr/ Found via codespell	4 months ago
Luz Paz	aa20bfcf99	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`	4 months ago
Luz Paz	d2bcdb71f8	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`	4 months ago
George Joseph	6cf62b6032	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	4 months ago
George Joseph	32b0b47348	.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`.	4 months ago
Mike Bradeen	a39f3d5adb	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	4 months ago
George Joseph	f4d881803a	.github: Refactor Releaser to use reusable workflow	4 months ago
George Joseph	a5d37e5ecb	.github: Change branch of reusable workflows to main.	4 months ago
George Joseph	1f56d5fe7e	.github: Refactor to use pull_request_target trigger. After careful review, we believe we can now use the "pull_request_target" workflow trigger instead of "pull_request" which required a separate privliged workflow to add labels and comments to PRs when they are submitted or updated. This allows us to greatly streamline our workflows and remove unneeded ones. * The OnPRChanged workflow was... * Renamed to OnPRCheck * Changed to trigger on pull_request_target and the "recheckpr" label. * Changed to simply call reusable workflows in asterisk-ci-actions. * Changed to use better concurrency groups. * The OnPRCPCheck and OnPRMergeApproved workflows were also... * Changed to simply call reusable workflows in asterisk-ci-actions. * Changed to use better concurrency groups. * The NightlyTest and CreateDocs were also tweaked	4 months ago
George Joseph	fdbb9e29dd	res_config_pgsql: Fix regression that removed dbname config. A recent commit accidentally removed the code that sets dbname. This commit adds it back in. Resolves: #1119	4 months ago
George Joseph	0b6a3df331	res_stir_shaken: Allow missing or anonymous CID to continue to the dialplan. The verification check for missing or anonymous callerid was happening before the endpoint's profile was retrieved which meant that the failure_action parameter wasn't available. Therefore, if verification was enabled and there was no callerid or it was "anonymous", the call was immediately terminated instead of giving the dialplan the ability to decide what to do with the call. * The callerid check now happens after the verification context is created and the endpoint's stir_shaken_profile is available. * The check now processes the callerid failure just as it does for other verification failures and respects the failure_action parameter. If set to "continue" or "continue_return_reason", `STIR_SHAKEN(0,verify_result)` in the dialplan will return "invalid_or_no_callerid". * If the endpoint's failure_action is "reject_request", the call will be rejected with `433 "Anonymity Disallowed"`. * If the endpoint's failure_action is "continue_return_reason", the call will continue but a `Reason: STIR; cause=433; text="Anonymity Disallowed"` header will be added to the next provisional or final response. Resolves: #1112	4 months ago
George Joseph	873c247f75	resource_channels.c: Fix memory leak in ast_ari_channels_external_media. Between ast_ari_channels_external_media(), external_media_rtp_udp(), and external_media_audiosocket_tcp(), the `variables` structure being passed around wasn't being cleaned up properly when there was a failure. * In ast_ari_channels_external_media(), the `variables` structure is now defined with RAII_VAR to ensure it always gets cleaned up. * The ast_variables_destroy() call was removed from external_media_rtp_udp(). * The ast_variables_destroy() call was removed from external_media_audiosocket_tcp(), its `endpoint` allocation was changed to to use ast_asprintf() as external_media_rtp_udp() does, and it now returns an error on failure. * ast_ari_channels_external_media() now checks the new return code from external_media_audiosocket_tcp() and sets the appropriate error response. Resolves: #1109	4 months ago
Holger Hans Peter Freyther	5e4fca062c	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.	4 months ago
George Joseph	00af359ea9	.github: Remove concurrency check in on-labelled workflows. Apparently you can't use `${{ github.event.number }}` in a concurrency block in a job that calls a reusable workflow. :(	4 months ago
Sean Bright	0883e5e44d	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.	4 months ago

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