* 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
* 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
* 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
* 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>
* 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
