* support kill dial if sd ep is media timeout
* support kill dial if sd ep is media timeout
* support kill dial if sd ep is media timeout
* add media timeout reason header to bye message
* wip
* wip
* make configuration for freeswitch media timeout
* make configuration for freeswitch media timeout
* wip
* 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
---------
Co-authored-by: Dave Horton <daveh@beachdognet.com>
* wip
* dont send if we have nothing to send
* initial testing with cartesia
* wip
---------
Co-authored-by: Hoan Luu Huu <110280845+xquanluu@users.noreply.github.com>
* writing alerts during startup and shutdown of feature-server
* feat/884: created constants for system component name and state
* feat/88: added 0.2.11 version of time-series
* feat/884: renamed constant, and added GracefulShutdownInProgress system alert
* 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>
* 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/893: made noResponseTimeout and noResponseGiveUpTimout independent
* support for giveUpActions implemented
* feat/902: using makeTask and exec of task to execute the giveUpActions
* feat/902: changed version of verb-specifications and speech-utils
* feat/902: fixed jslint errors
* feat/902: modified log
* feat/902: using a new event giveupWithTasks for processing giveUpActions
* feat/902: removed check for wakeupResolver and replaceApplication already taking care of wakeupResolver, also updated the verb-specifications version
* feat/902: removed sync for _onNoResponseGiveUpTimer function
* https://github.com/jambonz/jambonz-feature-server/issues/844
sending callSid in options, so that the callSid is sent to stt websocket in case of custom websocket
* feat/844: checking for existance of task.cs.callSid
* feat/844: changed the condition to task.cs?.callSid
* 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>
* fix bug where play incorrectly plays again after response received
* wip
* fix race condition where bot delay audio kcks off same instant we receive commands
