* 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
* refactor utilization jambonz app for queue and user
* refactor utilization jambonz app for queue and user
* refactor utilization jambonz app for queue and user
* refactor utilization jambonz app for queue and user
* refactor utilization jambonz app for queue and user
* feat: support app_json in application
* feat: support app_json in application
* update db schema to latest
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Co-authored-by: Quan HL <quanluuhoang8@gmail.com>
Co-authored-by: Dave Horton <daveh@beachdognet.com>
* 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>
* 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
