* 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
---------
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
---------
Co-authored-by: Rammohan Yadavalli <rammohan.yadavalli@kore.com>
* clear gather timeout if imterim result received
* fix to reset timeout timer if receive interrim result
* fix to reset timeout timer if receive interrim result
* feat/813 - notify speech-bargein-detected and dtmf-bargein-detected events
* fix for #826 race condition in say (#827)
* fix for #826 race condition in say
* wip
* wip
* wip
* wip
* wip
* wip
* wip
* wip
* Update transcription-utils.js (#802)
* Check the confidence levels of a transcript with minConfidence (#808)
* https://github.com/jambonz/jambonz-feature-server/issues/807
* feat/807: Using minConfidence from recognizer settings
* feat/807: new reason stt-min-confidence-error
* feat/807: sending stt-min-confidence instead of stt-min-confidence-error
* feat/807: sending stt-low-confidence instead of stt-min-confidence-error
* feat/807 - removed ? for this.data
* fix conference end is not sent when moderator leave conference (#825)
* fix conference end is not sent when moderator leave conference
* wip
* fix review comment
* feat/813: checking for playComplete before sending dtmf-bargein-detected event
* feat/813: added this.playComplete=true at the end of _killAudio method
* feat/813: removed empty line
* feat/813: removed nested if and added condition to main if
* feat/813: notifyStatus called when not playComplete
* feat/813: referring to time-series 0.2.9 version
* feat/813: generated package-lock.json
---------
Co-authored-by: Dave Horton <daveh@beachdognet.com>
Co-authored-by: Vinod Dharashive <vdharashive@gmail.com>
Co-authored-by: Hoan Luu Huu <110280845+xquanluu@users.noreply.github.com>
* 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
---------
Co-authored-by: Quan HL <quan.luuhoang8@gmail.com>
