* 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
* initial changes to use undici for http client and connection pooling
* use body.json() mixin
* logging
* add pipelining env var
* implement socket close
* bugfix: background gather for speech-only should still kill audio on dtmf entry when dtmfBargein is true
* initial changes for continuous asr
* move properties under recognizer
* update drachtio-srf@4.5.1
* catch exception on destroy
* 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
* 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
