* initial changes to support ibm watson
* update specs.json for ibm
* update to drachtio-fsmrf with support for ibm
* bugfix: set access token for ibm stt, not api_key
* fix name of api_key
* normalize ibm transcription results
* rework ibm credentials
* bugfix setting runtime speech creds
* bugfix: ibm region
* typo
* changes to transcribe for ibm watson
* implement connect handler
* bugfix: bind error
* proper use of result_index
* ibm error handling
* initial changes to support deepgram stt
* fixes for normalizing vendor-specific transcriptions
* update to latest drachtio-fsmrf with support for deepgram stt
* deepgram parsing error
* hints support for deepgram
* handling deepgram errors
* ignore late arriving transcripts for deepgram
* handling of empty transcripts
* transcribe changes
* allow deepgram stt credentials to be provided at run time
* bind channel in transcription handler
* fixes for transcribe when handling empty transcripts
* more empty transcript fixes
* update tests to latest modules
* add test cases for deepgram speech recognition
* 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
* gather/transcribe: support for azure custom speech models (endpoint id)
* allow azure stt custom speech endpoint id to be passed as property in recognizer
* fix to add custom stt endpoint to session speech credentials object
* 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
* initial changes for amd
* wip
* fix bug where transcripts were discarded
* a bit of refactoring, and adding support for avmd in config verb
* bug fixes
* initial changes to support siprec recording
* include additional params on SIP INFO to start recording
* add support for maniupulating recording via REST API
* fixes from testing pause/resume recording
* bugfix: background gather for speech-only should still kill audio on dtmf entry when dtmfBargein is true
* initial changes for continuous asr
* move properties under recognizer
* update drachtio-srf@4.5.1
* catch exception on destroy
* 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>
* 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
* 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
