Queue: Asterisk / FreeSWITCH PBX to Deepgram WebSocket streaming STT #70
Description
Integration: Asterisk / FreeSWITCH PBX + Deepgram Streaming STT
Origin
Requested in #67: "need more in‑depth documentation/guidance showing how to connect audio from PBX stacks (Asterisk + FreeSWITCH) into a WebSocket streaming STT connection (i.e., how to get RTP/PCM audio out of the PBX and into the Deepgram WS)."
What this should show
- How to capture RTP/PCM audio streams from Asterisk and/or FreeSWITCH
- How to forward that audio into a Deepgram WebSocket for live streaming speech-to-text
- Proper audio format configuration (sample rate, encoding, channels) for the PBX → Deepgram bridge
- A working end-to-end example: incoming call on PBX → real-time transcription via Deepgram
- Consider both Asterisk (ARI/AGI or AudioSocket) and FreeSWITCH (mod_audio_stream or ESL) approaches
Credentials likely needed
- DEEPGRAM_API_KEY
- Local Asterisk or FreeSWITCH instance for testing
Reference
- Deepgram FreeSWITCH partner page
- Deepgram streaming docs (WebSocket, audio formats, KeepAlive/CloseStream)
Queued by PM from #67 on 2026-03-30