* fixed dial transcribe is not able to received final transcribe when close call.
* wip
* fix review comment
* support call session delay detroy ep when current task is transcribe
* wip
* wip
* fixed review comments
* fixed review comments
* support kill dial if sd ep is media timeout
* support kill dial if sd ep is media timeout
* support kill dial if sd ep is media timeout
* add media timeout reason header to bye message
* wip
* wip
* make configuration for freeswitch media timeout
* make configuration for freeswitch media timeout
* wip
* fixes from testing with translator app
* more updates
* linting
* update gh actions to node 20
* add support for google v2 preconfigured recognizer
* add support for google voice activity events
* update to speech-utils@0.0.45
* update speech-utils to support caching azure tts
* transcribe must buffer transcripts for channel 1 and 2 separately
* further fix for accumulating transcripts
* linting
* deepgram sends transcripts with empty alternatives array
* fix deepgram returning an empty array
* initial support for coaching mode in conference
* wip
* wip
* add support for answer verb
* wip
* wip
* wip
* wip
* wip
* updates to rename option to dub
* wip
* wip
* wip
* update verb-specs
* wip
* wip
* wip
* wip
* wip
* wip
* wip
* wip
* add option to boost audio signal in main channel
* wip
* wip
* wip
* wip
* wip
* wip
* for now, bypass use of streaming apis when generating tts audio for dub tracks
* add nested dub to dial
* wip
* add support for filler noise
* kill filler noise when gather killed
* wip
* wip
* while using sayOnTrack, we have to enclose the say command in double quotes
* disableTtsStreaming = false
* allow transcribe of b leg only on dial verb
* dub.say can either be text or object like say verb with text and synthesizer
* remove loop for sayOnTrack
* update speech-utils
* fixes for testing transcribe verb and support for dub and boostAudioSignal in lcc commands
* add dial.boostAudioSignal
* fix bug where session-level recognizer settings incorrectly overwrite verb-level settings
* update verb specs
* update dial to support array of dub verbs
* fix bug setting gain
* lint
* wip
* update speech-utils
* use new endpoint methods for mod_dub
---------
Co-authored-by: Dave Horton <daveh@beachdognet.com>
* update to fsmrf with fix
* changes to support elevenlabs tts streaming
* say: add vendor data to span
* bug: tts spans must include cached property
* add env for JAMBONES_USE_FREESWITCH_TIMER_FD
* fix bug in prev commit
* wip
* linting
* wip - caching files generating by streaming tts
* wip caching
* cleanup some logs
* handle tts streaming failure, write alert
* update node version dependency
* set timerfd on outbound call scenarios
* default model to nova-2-phonecall when using deepgram
---------
Co-authored-by: Dave Horton <daveh@beachdognet.com>
* fix release freeswitch media properly
* if a leg is opus, modify b leg offer opus first
* if a leg is opus, modify b leg offer opus first
* wip
* wip
* fix review comments
* fix review comments
* fix review comments
* feat Audult call session should have its own requestor
* fix eslint
* fix eslint
* if user authenticate from http server instead of databse
* wip
* fix custom STT
* fix custom STT
* fix custom STT
* initial adds for otel tracing
* initial basic testing
* basic tracing for incoming calls
* linting
* add traceId to the webhook params
* trace webhook calls
* tracing: add new commands as tags when receiving async commands over websocket
* tracing new commands
* add summary for config verb
* trace async commands
* bugfix: undefined ref
* tracing: give time for final webhooks before closing root span
* tracing bugfix: span for background gather was not ended
* tracing - minor tag changes
* tracing - add span atttribute for reason call ended
* trace call status webhooks, add app version to trace output
* config: add support for automatically re-enabling
* env var to customize service name in tracing UI
* config: change to use 'sticky' attribute to re-enable bargein automatically
* fix warnings
* when adulting create a new root span
* when background gather triggers bargein via vad clear queue of tasks
* additional trace attributes for dial and refer
* fix dial tracing
* add better summary for dial
* fix prev commit
* add exponential backoff to WsRequestor reconnection logic
* add calling number to log metadata, as this will be frequently the key data given for troubleshooting
* add accountSid to log metadata
* make handshake timeout for ws connections configurable with default 1.5 secs
* rename env var
* fix bug prev checkin
* logging fixes
* consistent env naming
* Dial: handle incoming REFER on either leg by calling referHook, if configured
* lint
* modify payload of referHook
* support target.trunk on rest createCall api
* bugfix: gather partial result hook was not working
* lint
* handling of incoming REFER
* JAMBONES_NETWORK_CIDR not needed for K8S
* fix bug setting fsUUID in K8S scenario
* bugfix: dial music was not stopped when a dial verb times out (#56)
* initial WIP to remove freeswitch from media path when not recording or transcribing dial calls
* implement release-media and anchor-media operations
* mute/unmute now handled by rtpengine
* Dial: dtmf detection now based on SIP INFO events from sbcs and rtpengine
* add reason to gather action, bugfixes for transcribe and say
