* wip
* add TtsStreamingBuffer class to abstract handling of streaming tokens
* wip
* add throttling support
* support background ttsStream (#995)
* wip
* add TtsStreamingBuffer class to abstract handling of streaming tokens
* wip
* support background ttsStream
* wip
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Co-authored-by: Dave Horton <daveh@beachdognet.com>
* wip
* dont send if we have nothing to send
* initial testing with cartesia
* wip
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Co-authored-by: Hoan Luu Huu <110280845+xquanluu@users.noreply.github.com>
* Fix the issue for outbound calls that always the None credentials were used. session:new for rest dial did not contain recognizer.label and synthesizer.label
* update comment
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Co-authored-by: mfrindt <m.frindt@cognigy.com>
* feat/975: fixed continuous asr not stopping when asrDtmfTerminationDigit is configured
* feat/975: giving first preference to asrDtmfTerminationDigit if there is already ASR happened
* fix transcribe fixes for speechmatics
* update to verb-specs with fixes for speechmatics
* add support for speechmatics translation
* add handlers for receiving translations
* call translation hookd
* gather: no need to restart speechmatics after a final transcript during continuous asr
* graceful shutdown
* wip
* wip
* wip
* wip
* wip
* feature server should send USER call to the sbc sip that is connect with the user
* feature server should send USER call to the sbc sip that is connect with the user
* feature server should send USER call to the sbc sip that is connect with the user
* fix review comment
* add env variable to enable the feature
* add env variable to enable the feature
* add env variable to enable the feature
* minor test update
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Co-authored-by: Dave Horton <daveh@beachdognet.com>
* include X-CID for dial outbound if the call-session is outbound
* include X-CID for dial outbound if the call-session is outbound
* include X-CID for dial outbound if the call-session is outbound
* include X-CID for dial outbound if the call-session is outbound
* add support for aws language model name when transcribing
* wip - from prev branch
* wip
* support both aws grpc and ws api - detect based on transcription payload
* update to drachtio-fsmrf@4.0.0
* fix for grpc compatibility, requires JAMBONES_AWS_TRANSCRIBE_USE_GRPC env
* back out major update to drachtio-srf and fsmrf; that should come in a separate PR
* update drachtio-srf
* allow move to next task if say verb is failed because of speech credential
* allow move to next task if say verb is failed because of speech credential
* allow move to next task if say verb is failed because of speech credential
* wip
* wip
* feat/864: checking for undefined, because 0 is a valid value for minBargeinWordCount
* feat/864: checking for undefined and null
* feat/864: corrected spelling of mode and added check for undefined as 0 is a valid value for vad.mode
* fix#883 that after kicked from conference, no long receive freeswitch CUSTOM event
* fix#883 that after kicked from conference, no long receive freeswitch CUSTOM event
* reset Esl Custom event after conference.
* update drachtio fsmrf version
* feat/868: Use the properties from global config in verb for TTS
* feat/868: setting this.options to combination of cs.synthesizer.options and this.options
* feat/868: Move the logic of copying cs properties to parent class tts-task.js
* feat/868: add empty line that was removed, say.js restored to original version
* feat/868: moved _synthesizeWithSpecificVendor to tts-task.js
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Co-authored-by: Rammohan Yadavalli <rammohan.yadavalli@kore.com>
