Queue: Asterisk / FreeSWITCH PBX to Deepgram WebSocket streaming STT #70

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