* feat/836: capturing callSid for STT and TTS alerts
* feat/836: corrected assignment of callSid and added target_sid at few more alerts
* update github action
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Co-authored-by: Quan HL <quan.luuhoang8@gmail.com>
* Aws polly engine fix
engine parameter was not able to change using synthesizer
* WIP
code correction and set default engine to Neural
* WIP
* WIP
Updated tts-task.js
* WIP
* kill play task if bot responds verbs while actionHook delay is enabled (#712)
* kill play task if bot responds verbs while actionHook delay is enabled
* fix actionHook delay continues even the bot already responded verbs
* wip
* wip
* wip
* gather is hang if listenDuringPrompt = false and say/play task throw exception (#717)
* merge fix for Support ASR TTS fallback (#713)
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Co-authored-by: Hoan Luu Huu <110280845+xquanluu@users.noreply.github.com>
* 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
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Co-authored-by: Dave Horton <daveh@beachdognet.com>
* update speech util to support whisper stream
* minor editing of span attributes
* more span attrs cleanup
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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
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Co-authored-by: Dave Horton <daveh@beachdognet.com>
* Update say task and add possibility to use elevenlabs options from synthesizer
* revert ms change
* fix contdition for alerting
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Co-authored-by: Markus Frindt <m.frindt@cognigy.com>
* initial changes to gather to support nuance stt
* updateSpeechCredentialLastUsed could be called without a speech_credential_sid if credentials are passed in the flow
* fix bugname
* typo
* added handlers for nuance
* logging
* major refactor of parsing transcriptions
* initial support for nuance in transcribe verb
* updates from testing
* cleanup some tests
* update action
* typo
* gather: start nuance timers after say/play completes
* update drachtio-fsrmf
* refactor some code
* typo
* log nuance error detail
* timeout handling
* typo
* handle nuance 413 response when recognition times out
* typo in specs.json
* add support for nuance resources
* fixes and tests for transcribe
* remove logging from test
* initial support for kryptonEndpoint
* try getting access token even when using krypton
* typo in kryptonEndpoint property
* add support for Nuance tts
* parse nuance voice and model for tts
* use nuance credentials from db
* update to db-helpers@0.7.0 with caching option
* add support for azure audio logging in gather/transcribe
* sync package-lock.json
* typo for media bug name in azure and punctuation fix
* say: split very long text intelligently
* more fixes from testing
* update to latest synthAudio
* fixes from testing
* modify Task#exec to take resources as an object rather than argument list
* pass 2 endpoints to Transcribe when invoked in a SipRec call session
* logging
* change siprec invite to sendrecv just so freeswitch does not try to reinvite (TODO: block outgoing media at rtpengine)
* Config: when enabling recording, block until siprec dialog is established
* missed play verb in commit 031c79d
* linting
* bugfix: get final transcript in siprec call
* add b3 header for trace propagation on initial webhook
* logging
* add tracing context to all webhooks
* Add span parameter to Task.getTracingPropagation. Pass proper span to getTracingPropagation calls in Task methods to propagate the proper spanId (#91)
* some tracing cleanup
* bugfix: azure stt results need to be ordered by confidence level before processing (#92)
* fix assertion
* bugfix: vad was not enabled on config verb, restart STT on empty transcript in gather
* gather: dont send webhook if call is gone
* rest outdial: handle 302 redirect so we can later cancel request if needed (#95)
* gather: restart if we get an empty transcript (looking at you, Azure)
Co-authored-by: javibookline <98887695+javibookline@users.noreply.github.com>
* 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
* remove cognigy verb
* initial implementation of config verb
* further updates to config
* Bot mode alex (#75)
* do not use default as value for TTS/STT
* fix gather listener if no say or play provided
Co-authored-by: akirilyuk <a.kirilyuk@cognigy.com>
* gather: listenDuringPrompt requires a nested play/say
* fix exception
* say: fix exception where caller hangs up during say
* bugfix: sip refer was not ending if caller hungup during refer
* add support for sip:request to ws commands
* gather: when bargein is set and minBargeinWordCount is zero, kill audio on endOfUtterrance
* gather/transcribe: add support for google boost and azure custom endpoints
* minor logging changes
* lint error
Co-authored-by: akirilyuk <45361199+akirilyuk@users.noreply.github.com>
Co-authored-by: akirilyuk <a.kirilyuk@cognigy.com>
* add bargein support to gather
* bugfix: gather handles interim results from azure
* gather: support for min/max digits and interdigit timeout
* add task summary to some log messages
* logging improvements
* 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
