asterisk

Commit Graph

Author	SHA1	Message	Date
Naveen Albert	60219d49c9	chan_iax2: Add CHANNEL getter to retrieve auth method. Add a property to the CHANNEL method to retrieve the auth method, which can be used to retrieve the specific auth method actually negotiated for a call (e.g. RSA, MD5, etc.). Also clean up some of the documentation for the secure properties to clarify how these relate to call encryption. Resolves: #1878 UserNote: CHANNEL(auth_method) can now be used to retrieve the auth method negotiated for a call on IAX2 channels.	5 months ago
Naveen Albert	eadacd4a4d	chan_iax2: Add another check to abort frame handling if datalen < 0. Commit `2da221e217` added a missing abort if datalen < 0 check on a code path and an assertion inside iax_frame_wrap if we ever encountered a frame with a negative frame length (which will eventually cause a crash). Add another missing abort check for negative datalen, exposed by this assertion. (Similar to the previous commit, this is a video frame with a datalen of -1). Resolves: #1865	5 months ago
George Joseph	041a183bac	chan_pjsip: Set correct cause codes for non-2XX responses. Redirects initiated by 302 response codes weren't handled correctly when setting the hangup cause code and tech cause code on the responding channel. They're now set to 23 (REDIRECTED_TO_NEW_DESTINATION) and 302 (Moved permanently). Other non-2XX response codes also had issues. A new API ast_channel_dialed_causes_iterator() was added to retrieve the hangup cause codes for a channel. chan_pjsip_session_end() in chan_pjsip has been refactored to set the correct cause codes on a channel based on the cause codes added by chan_pjsip_incoming_response_update_cause(). Copious amounts of debugging and comments were also added. Resolves: #1819	6 months ago
Ben Ford	3c02b8bd3d	chan_websocket_doc.xml: Add d(media_direction) option. Adds documentation for the 'd' option to set media direction for chan_websocket.	7 months ago
George Joseph	2f01334a19	chan_websocket: Remove silence generation and frame padding. The original chan_websocket implementation attempted to improve the call quality experience by generating silence frames to send to the core when no media was being read from the websocket and padding frames with silence when short frames were read from the websocket. Both of these required switching the formats on the channel to slin for short periods of time then switching them back to whatever format the websocket channel was configured for. Unfortunately, the format switching caused issues when transcoding was required and the transcode_via_sln option was enabled in asterisk.conf (which it is by default). The switch would cause the transcoding path to be incorrectly set resulting in malformed RTP packets being sent back out to the caller on the other end which the caller heard as loud noise. After looking through the code and performing multiple listening tests, the best solution to this problem was to just remove the code that was attempting to generate the silence. There was no decrease in call quality whatsoever and the code is a bit simpler. None of the other channel drivers nor res_rtp_asterisk generate silence frames or pad short frames which backed up decision. Resolves: #1785	7 months ago
Ben Ford	3e5ad78d28	chan_websocket: Add media direction. Currently, WebSockets both accept and send media without the option to disable one or the other. This commit adds the ability to set the media direction for a WebSocket, making it unidirectional or bidirectional (the default). Direction is done from the point of view of the application, NOT Asterisk. The allowed values are 'both', 'in', and 'out'. If media direction is 'both' (the default), Asterisk accepts and sends media to the application. If it is 'in', Asterisk will drop any media received from the application. If it is 'out', Asterisk will not write any media frames to the application. UserNote: WebSocket now supports media direction, allowing for unidirectional media. This is done from the perspective of the application and can be set via channel origination, external media, or commands sent from the application. Check out https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket/ for more.	7 months ago
Naveen Albert	23fb0524ca	chan_dahdi: Fix discarded-qualifiers errors. Fix discarded-qualifiers errors to compile successfully with gcc 15.2.1. Associated changes have also been made to libss7; however, for compatibility we cast const char* values to char*. In the future, these casts could be removed. Resolves: #1786	7 months ago
Naveen Albert	ad7be5c675	build: Fix unused-but-set-variable warnings with gcc 16. Fix or remove a few variables that were being set but not actually used anywhere, causing warnings with gcc 16. Resolves: #1783	7 months ago
Naveen Albert	aed6b5f9ee	chan_iax2: Fix crash due to negative length frame lengths. chan_iax2 has several code paths where a frame's data length is calculated by subtraction. On some paths, there is a check for negative length. One of these paths is missing this check, and on this path, it is possible for the result to be negative, leading to a crash as a result of memory operations using the bogus length. Add a check to capture this off-nominal case. This will log the appropriate warnings as in other cases and prevent a crash. Also update the log messages to be clearer. Resolves: #1707	7 months ago
Joshua C. Colp	7aba01a49d	build: Fix GCC discarded-qualifiers const errors. GCC 15.2.1 pays attention to the discarding of the const qualifier when strchr, strrchr, memchr, or memrchr are now used. This change fixes numerous errors with this throughout the tree. The fixes can be broken down into the following: 1. The return value should be considered const. 2. The value passed to strchr or strrchr can be cast as it is expected and allowed to be modified. 3. The pointer passed to strchr or strrchr is not meant to be modified and so the contents must be duplicated. 4. It was declared const and never should have been.	7 months ago
Joshua C. Colp	af3ca1b6cd	endpoints: Allow access to latest snapshot directly. This change adds an API call to allow direct access to the latest snapshot of an ast_endpoint. This is then used by chan_pjsip when calculating device state, eliminating the need to access the cache which would incur a container find and access.	7 months ago
Joe Garlick	4ab499195e	chan_websocket: Fixed Ping/Pong messages hanging up the websocket channel When chan_websocket received a Ping or a Pong opcode it would cause the channel to hangup. This change allows Ping/Pong opcodes and allows them to silently pass	8 months ago
George Joseph	f8236f13e1	chan_websocket: Use the channel's ability to poll fds for the websocket read. We now add the websocket's file descriptor to the channel's fd array and let it poll for data availability instead if having a dedicated thread that does the polling. This eliminates the thread and allows removal of most explicit locking since the core channel code will lock the channel to prevent simultaneous calls to webchan_read, webchan_hangup, etc. While we were here, the hangup code was refactored to use ast_hangup_with_cause instead of directly queueing an AST_CONTROL_HANGUP frame. This allows us to set hangup causes and generate snapshots. For a bit of extra debugging, a table of websocket close codes was added to http_websocket.h with an accompanying "to string" function added to res_http_websocket.c Resolves: #1683	9 months ago
Joshua C. Colp	a12251ca88	pjsip: Move from threadpool to taskpool This change moves the PJSIP module from the threadpool API to the taskpool API. PJSIP-specific implementations for task usage have been removed and replaced with calls to the optimized taskpool implementations instead. The need for a pool of serializers has also been removed as taskpool inherently provides this. The default settings have also been changed to be more realistic for common usage. UpgradeNote: The threadpool_* options in pjsip.conf have now been deprecated though they continue to be read and used. They have been replaced with taskpool options that give greater control over the underlying taskpool used for PJSIP. An alembic upgrade script has been added to add these options to realtime as well.	9 months ago
phoneben	bb008fbf7c	Disable device state caching for ephemeral channels chan_audiosocket/chan_rtp/res_stasis_snoop: Disable device state caching for ephemeral channels Resolves: #1638	9 months ago
George Joseph	1bb03370f4	chan_websocket: Add locking in send_event and check for NULL websocket handle. On an outbound websocket connection, when the triggering caller hangs up, webchan_hangup() closes the outbound websocket session and sets the websocket session handle to NULL. If the hangup happened in the tiny window between opening the outbound websocket connection and before read_thread_handler() was able to send the MEDIA_START message, it could segfault because the websocket session handle was NULL. If it didn't actually segfault, there was also the possibility that the websocket instance wouldn't get cleaned up which could also cause the channel snapshot to not get cleaned up. That could cause memory leaks and `core show channels` to list phantom WebSocket channels. To prevent the race, the send_event() macro now locks the websocket_pvt instance and checks the websocket session handle before attempting to send the MEDIA_START message. Resolves: #1643 Resolves: #1645	9 months ago
Sean Bright	346474635f	chan_websocket: Fix crash on DTMF_END event. Resolves: #1604	10 months ago
Joe Garlick	2d11bb044e	chan_websocket.c: Tolerate other frame types Currently, if chan_websocket receives an un supported frame like comfort noise it will exit the websocket. The proposed change is to tolerate the other frames by not sending them down the websocket but instead just ignoring them. Resolves: #1587	10 months ago
George Joseph	72548c17ab	chan_websocket: Add ability to place a MARK in the media stream. Also cleaned up a few unused #if blocks, and started sending a few ERROR events back to the apps. Resolves: #1574 DeveloperNote: Apps can now send a `MARK_MEDIA` command with an optional `correlation_id` parameter to chan_websocket which will be placed in the media frame queue. When that frame is dequeued after all intervening media has been played to the core, chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the app with the same correlation_id (if any).	11 months ago
George Joseph	6b45c0940e	chan_websocket: Add capability for JSON control messages and events. With recent enhancements to chan_websocket, the original plain-text implementation of control messages and events is now too limiting. We probably should have used JSON initially but better late than never. Going forward, enhancements that require control message or event changes will only be done to the JSON variants and the plain-text variants are now deprecated but not yet removed. * Added the chan_websocket.conf config file that allows setting which control message format to use globally: "json" or "plain-text". "plain-text" is the default for now to preserve existing behavior. * Added a dialstring option `f(json\|plain-text)` to allow the format to be overridden on a call-by-call basis. Again, 'plain-text' is the default for now to preserve existing behavior. The JSON for commands sent by the app to Asterisk must be... `{ "command": "<command>" ... }` where `<command>` is one of `ANSWER`, `HANGUP`, `START_MEDIA_BUFFERING`, etc. The `STOP_MEDIA_BUFFERING` command takes an additional, optional parameter to be returned in the corresponding `MEDIA_BUFFERING_COMPLETED` event: `{ "command": "STOP_MEDIA_BUFFERING", "correlation_id": "<correlation id>" }`. The JSON for events sent from Asterisk to the app will be... `{ "event": "<event>", "channel_id": "<channel_id>" ... }`. The `MEDIA_START` event will now look like... ``` { "event": "MEDIA_START", "connection_id": "media_connection1", "channel": "WebSocket/media_connection1/0x5140001a0040", "channel_id": "1761245643.1", "format": "ulaw", "optimal_frame_size": 160, "ptime": 20, "channel_variables": { "DIALEDPEERNUMBER": "media_connection1/c(ulaw)", "MEDIA_WEBSOCKET_CONNECTION_ID": "media_connection1", "MEDIA_WEBSOCKET_OPTIMAL_FRAME_SIZE": "160" } } ``` Note the addition of the channel variables which can't be supported with the plain-text formatting. The documentation will be updated with the exact formats for all commands and events. Resolves: #1546 Resolves: #1563 DeveloperNote: The chan_websocket plain-text control and event messages are now deprecated (but remain the default) in favor of JSON formatted messages. See https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket for more information. DeveloperNote: A "transport_data" parameter has been added to the channels/externalMedia ARI endpoint which, for websocket, allows the caller to specify parameters to be added to the dialstring for the channel. For instance, `"transport_data": "f(json)"`.	11 months ago
Joshua C. Colp	ff56cec9a1	Revert "pjsip: Move from threadpool to taskpool" This reverts commit `bb6b76c2d8`.	11 months ago
George Joseph	75d74ff900	chan_pjsip: Disable SSRC change for WebRTC endpoints. Commit `b333ee3b` introduced a fix to chan_pjsip that addressed RTP issues with blind transfers and some SBCs. Unfortunately, the fix broke some WebRTC clients that are sensitive to SSRC changes and non-monotonic timestamps so the fix is now disabled for endpoints with the "bundle" parameter set to true. Resolves: #1535	11 months ago
gauravs456	7795cc150b	chan_websocket: Add channel_id to MEDIA_START, DRIVER_STATUS and DTMF_END events. Resolves: #1544	11 months ago
Joshua C. Colp	8d39faaf12	pjsip: Move from threadpool to taskpool This change moves the PJSIP module from the threadpool API to the taskpool API. PJSIP-specific implementations for task usage have been removed and replaced with calls to the optimized taskpool implementations instead. The need for a pool of serializers has also been removed as taskpool inherently provides this. The default settings have also been changed to be more realistic for common usage. UpgradeNote: The threadpool_* options in pjsip.conf have now been deprecated though they continue to be read and used. They have been replaced with taskpool options that give greater control over the underlying taskpool used for PJSIP. An alembic upgrade script has been added to add these options to realtime as well.	11 months ago
George Joseph	72e9e4665e	chan_pjsip: Add technology-specific off-nominal hangup cause to events. Although the ISDN/Q.850/Q.931 hangup cause code is already part of the ARI and AMI hangup and channel destroyed events, it can be helpful to know what the actual channel technology code was if the call was unsuccessful. For PJSIP, it's the SIP response code. * A new "tech_hangupcause" field was added to the ast_channel structure along with ast_channel_tech_hangupcause() and ast_channel_tech_hangupcause_set() functions. It should only be set for off-nominal terminations. * chan_pjsip was modified to set the tech hangup cause in the chan_pjsip_hangup() and chan_pjsip_session_end() functions. This is a bit tricky because these two functions aren't always called in the same order. The channel that hangs up first will get chan_pjsip_session_end() called first which will trigger the core to call chan_pjsip_hangup() on itself, then call chan_pjsip_hangup() on the other channel. The other channel's chan_pjsip_session_end() function will get called last. Unfortunately, the other channel's HangupRequest events are sent before chan_pjsip has had a chance to set the tech hangupcause code so the HangupRequest events for that channel won't have the cause code set. The ChannelDestroyed and Hangup events however will have the code set for both channels. * A new "tech_cause" field was added to the ast_channel_snapshot_hangup structure. This is a public structure so a bit of refactoring was needed to preserve ABI compatibility. * The ARI ChannelHangupRequest and ChannelDestroyed events were modified to include the "tech_cause" parameter in the JSON for off-nominal terminations. The parameter is suppressed for nominal termination. * The AMI SoftHangupRequest, HangupRequest and Hangup events were modified to include the "TechCause" parameter for off-nominal terminations. Like their ARI counterparts, the parameter is suppressed for nominal termination. DeveloperNote: A "tech_cause" parameter has been added to the ChannelHangupRequest and ChannelDestroyed ARI event messages and a "TechCause" parameter has been added to the HangupRequest, SoftHangupRequest and Hangup AMI event messages. For chan_pjsip, these will be set to the last SIP response status code for off-nominally terminated calls. The parameter is suppressed for nominal termination.	11 months ago
George Joseph	b84dfe94f5	chan_websocket.c: Change payload references to command instead. Some of the tests in process_text_message() were still comparing to the websocket message payload instead of the "command" string. Resolves: #1525	12 months ago
Naveen Albert	e8d97e5d60	sig_analog: Allow '#' to end the inter-digit timeout when dialing. It is customary to allow # to terminate digit collection immediately when there would normally be a timeout. However, currently, users are forced to wait for the timeout to expire when dialing numbers that are prefixes of other valid matches, and there is no way to end the timeout early. Customarily, # terminates the timeout, but at the moment, this is just rejected unless there happens to be a matching extension ending in #. Allow # to terminate the timeout in cases where there is no dialplan match. This ensures that the dialplan is always respected, but if a valid extension has been dialed that happens to prefix other valid matches, # can be used to dial it immediately. Resolves: #1510	12 months ago
Naveen Albert	683e2b435a	sig_analog: Eliminate potential timeout with Last Number Redial. If Last Number Redial is used to redial, ensure that we do not wait for further digits. This was possible if the number that was last dialed is a prefix of another possible dialplan match. Since all we did is copy the number into the extension buffer, if other matches are now possible, there would thus be a timeout before the call went through. We now complete redialed calls immediaetly in all cases. Resolves: #1483	1 year ago
Bastian Triller	573c8faa09	Fix some doxygen, typos and whitespace	1 year ago
George Joseph	20a6531458	chan_websocket: Fix codec validation and add passthrough option. * Fixed an issue in webchan_write() where we weren't detecting equivalent codecs properly. * Added the "p" dialstring option that puts the channel driver in "passthrough" mode where it will not attempt to re-frame or re-time media coming in over the websocket from the remote app. This can be used for any codec but MUST be used for codecs that use packet headers or whose data stream can't be broken up on arbitrary byte boundaries. In this case, the remote app is fully responsible for correctly framing and timing media sent to Asterisk and the MEDIA text commands that could be sent over the websocket are disabled. Currently, passthrough mode is automatically set for the opus, speex and g729 codecs. * Now calling ast_set_read_format() after ast_channel_set_rawreadformat() to ensure proper translation paths are set up when switching between native frames and slin silence frames. This fixes an issue with codec errors when transcode_via_sln=yes. Resolves: #1462	1 year ago
Naveen Albert	8ff320ea4e	chan_dahdi: Add DAHDI_CHANNEL function. Add a dialplan function that can be used to get/set properties of DAHDI channels (as opposed to Asterisk channels). This exposes properties that were not previously available, allowing for certain operations to now be performed in the dialplan. Resolves: #1455 UserNote: The DAHDI_CHANNEL function allows for getting/setting certain properties about DAHDI channels from the dialplan.	1 year ago
Joe Garlick	219ec0141c	chan_websocket.c: Add DTMF messages Added DTMF messages to the chan_websocket feature. When a user presses DTMF during a call over chan_websocket it will send a message like: "DTMF_END digit:1" Resolves: https://github.com/asterisk/asterisk-feature-requests/issues/70	1 year ago
Naveen Albert	54cab33a43	sig_analog: Skip Caller ID spill if usecallerid=no. If Caller ID is disabled for an FXS port, then we should not send any Caller ID spill on the line, as we have no Caller ID information that we can/should be sending. Resolves: #1394	1 year ago
Naveen Albert	576a29b988	chan_dahdi: Fix erroneously persistent dialmode. It is possible to modify the dialmode setting in the chan_dahdi/sig_analog private using the CHANNEL function, to modify it during calls. However, it was not being reset between calls, meaning that if, for example, tone dialing was disabled, it would never work again unless explicitly enabled. This fixes the setting by pairing it with a "perm" version of the setting, as a few other features have, so that it can be reset to the permanent setting between calls. The documentation is also clarified to explain the interaction of this setting and the digitdetect setting more clearly. Resolves: #1378	1 year ago
George Joseph	5d822d64ef	chan_websocket: Allow additional URI parameters to be added to the outgoing URI. * Added a new option to the WebSocket dial string to capture the additional URI parameters. * Added a new API ast_uri_verify_encoded() that verifies that a string either doesn't need URI encoding or that it has already been encoded. * Added a new API ast_websocket_client_add_uri_params() to add the params to the client websocket session. * Added XML documentation that will show up with `core show application Dial` that shows how to use it. Resolves: #1352 UserNote: A new WebSocket channel driver option `v` has been added to the Dial application that allows you to specify additional URI parameters on outgoing connections. Run `core show application Dial` from the Asterisk CLI to see how to use it.	1 year ago
George Joseph	09802fbef2	chan_websocket: Fix buffer overrun when processing TEXT websocket frames. ast_websocket_read() receives data into a fixed 64K buffer then continually reallocates a final buffer that, after all continuation frames have been received, is the exact length of the data received and returns that to the caller. process_text_message() in chan_websocket was attempting to set a NULL terminator on the received payload assuming the payload buffer it received was the large 64K buffer. The assumption was incorrect so when it tried to set a NULL terminator on the payload, it could, depending on the state of the heap at the time, cause heap corruption. process_text_message() now allocates its own payload_len + 1 sized buffer, copies the payload received from ast_websocket_read() into it then NULL terminates it prevent the possibility of the overrun and corruption. Resolves: #1384	1 year ago
Naveen Albert	6f2167989a	sig_analog: Fix SEGV due to calling strcmp on NULL. Add an additional check to guard against the channel application being NULL. Resolves: #1380	1 year ago
Naveen Albert	8ef3e4fb3d	sig_analog: Properly handle STP, ST2P, and ST3P for fgccamamf. Previously, we were only using # (ST) as a terminator, and not handling A (STP), B (ST2P), or C (ST3P), which erroneously led to it being treated as part of the dialed number. Parse any of these as the start digit. Resolves: #1301	1 year ago
kodokaii	f9219284cb	chan_websocket: Reset frame_queue_length to 0 after FLUSH_MEDIA In the WebSocket channel driver, the FLUSH_MEDIA command clears all frames from the queue but does not reset the frame_queue_length counter. As a result, the driver incorrectly thinks the queue is full after flushing, which prevents new multimedia frames from being sent, especially after multiple flush commands. This fix sets frame_queue_length to 0 after flushing, ensuring the queue state is consistent with its actual content. Fixes: #1304	1 year ago
Martin Tomec	9e7788ad21	chan_pjsip.c: Change SSRC after media source change When the RTP media source changes, such as after a blind transfer, the new source introduces a discontinuous timestamp. According to RFC 3550, Section 5.1, an RTP stream's timestamp for a given SSRC must increment monotonically and linearly. To comply with the standard and avoid a large timestamp jump on the existing SSRC, a new SSRC is generated for the new media stream. This change resolves known interoperability issues with certain SBCs (like Sonus/Ribbon) that stop forwarding media when they detect such a timestamp violation. This code uses the existing implementation from chan_sip. Resolves: #927	1 year ago
Naveen Albert	415daae95f	users.conf: Remove deprecated users.conf integration. users.conf was deprecated in Asterisk 21 and is now being removed for Asterisk 23, in accordance with the Asterisk deprecation policy. This consists of: * Removing integration with app_directory, app_voicemail, chan_dahdi, chan_iax2, and AMI. * users.conf was also partially used for res_phoneprov, and this remaining functionality is consolidated to a separate phoneprov_users.conf, used only by res_phoneprov. Resolves: #1292 UpgradeNote: users.conf has been removed and all channel drivers must be configured using their specific configuration files. The functionality previously in users.conf for res_phoneprov is now in phoneprov_users.conf.	1 year ago
George Joseph	5963e624e2	Media over Websocket Channel Driver * Created chan_websocket which can exchange media over both inbound and outbound websockets which the driver will frame and time. See http://s.asterisk.net/mow for more information. * res_http_websocket: Made defines for max message size public and converted a few nuisance verbose messages to debugs. * main/channel.c: Changed an obsolete nuisance error to a debug. * ARI channels: Updated externalMedia to include chan_websocket as a supported transport. UserNote: A new channel driver "chan_websocket" is now available. It can exchange media over both inbound and outbound websockets and will both frame and re-time the media it receives. See http://s.asterisk.net/mow for more information. UserNote: The ARI channels/externalMedia API now includes support for the WebSocket transport provided by chan_websocket.	1 year ago
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	1 year ago
Naveen Albert	2ced79259a	sig_analog: Add Call Waiting Deluxe support. Adds support for Call Waiting Deluxe options to enhance the current call waiting feature. As part of this change, a mechanism is also added that allows a channel driver to queue an audio file for Dial() to play, which is necessary for the announcement function. ASTERISK-30373 #close Resolves: #271 UserNote: Call Waiting Deluxe can now be enabled for FXS channels by enabling its corresponding option.	1 year ago
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	1 year 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,,#...).	2 years 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>	2 years 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	2 years 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`	2 years ago
Holger Hans Peter Freyther	9fc631d028	ari/pjsip: Make it possible to control transfers through ARI Introduce a ChannelTransfer event and the ability to notify progress to ARI. Implement emitting this event from the PJSIP channel instead of handling the transfer in Asterisk when configured. Introduce a dialplan function to the PJSIP channel to switch between the "core" and "ari-only" behavior. UserNote: Call transfers on the PJSIP channel can now be controlled by ARI. This can be enabled by using the PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.	2 years ago

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8806 Commits (65b98d46bc07701ca414abcd897fbe8a96bec769)