350-asterisk-freeswitch-deepgram-stt-python

Asterisk / FreeSWITCH PBX to Deepgram Streaming STT

Bridge real-time phone call audio from Asterisk or FreeSWITCH into Deepgram's live speech-to-text API. This example shows how to capture RTP/PCM audio from a PBX and stream it over a WebSocket to Deepgram for real-time transcription.

What you'll build

A Python WebSocket server that acts as a bridge between your PBX and Deepgram. Incoming calls on Asterisk (via AudioSocket) or FreeSWITCH (via mod_audio_stream) send their audio to this server, which forwards it to Deepgram's streaming STT API and prints live transcripts to the console.

Prerequisites

• Python 3.11+
• Deepgram account — get a free API key
• Asterisk 16+ with app_audiosocket module, or FreeSWITCH with mod_audio_stream

Environment variables

Variable	Where to find it
DEEPGRAM_API_KEY	Deepgram console

Copy .env.example to .env and fill in your values.

Install and run

cd examples/260-asterisk-freeswitch-deepgram-stt-python

pip install -r requirements.txt

cp .env.example .env
# Edit .env and add your DEEPGRAM_API_KEY

python src/bridge.py

The bridge listens on ws://0.0.0.0:8765 by default. Use --port to change it.

Asterisk dialplan configuration

Add to your Asterisk dialplan (extensions.conf) to route call audio to the bridge:

[transcribe]
exten => _X.,1,Answer()
 same => n,AudioSocket(ws://bridge-host:8765/asterisk)
 same => n,Hangup()

Asterisk AudioSocket sends signed-linear 16-bit PCM at 8 kHz mono by default. The bridge parses AudioSocket's TLV framing (type-length-value) to extract audio frames.

FreeSWITCH dialplan configuration

Add to your FreeSWITCH dialplan to stream call audio to the bridge:

<action application="answer"/>
<action application="audio_stream" data="ws://bridge-host:8765/freeswitch 16000 mono L16"/>

FreeSWITCH mod_audio_stream sends raw PCM frames directly — no framing protocol, just binary audio on the WebSocket.

Key parameters

Parameter	Value	Description
model	nova-3-phonecall	Deepgram model optimised for telephony audio (8/16 kHz)
encoding	linear16	Signed 16-bit little-endian PCM — the native format of both PBX platforms
sample_rate	8000 / 16000	8 kHz for Asterisk default, 16 kHz for FreeSWITCH (higher = better accuracy)
smart_format	True	Adds punctuation, capitalisation, and number formatting
interim_results	True	Returns partial transcripts while the caller is still speaking
utterance_end_ms	1000	Fires an utterance-end event after 1 second of silence

How it works

1. PBX receives a call — Asterisk or FreeSWITCH answers and is configured to stream audio to this bridge
2. Audio reaches the bridge — Asterisk sends AudioSocket TLV frames to /asterisk; FreeSWITCH sends raw PCM to /freeswitch
3. Bridge opens a Deepgram connection — using the Python SDK's client.listen.v1.connect() with telephony-optimised settings
4. Audio is forwarded — each PCM chunk is sent to Deepgram via connection.send_media()
5. Transcripts arrive — Deepgram fires EventType.MESSAGE callbacks with interim and final transcripts, which the bridge logs to the console
6. Call ends — the PBX closes the WebSocket; the bridge sends close_stream to Deepgram

Architecture

Phone Call
    |
    | RTP audio
    v
Asterisk / FreeSWITCH PBX
    |
    | WebSocket (AudioSocket TLV or raw PCM)
    v
bridge.py (this server)
    |
    | Deepgram Python SDK (WebSocket)
    v
Deepgram Live STT (nova-3-phonecall)
    |
    | transcript events
    v
Console output (or your application)

Related

• Deepgram FreeSWITCH integration
• Deepgram Live STT docs
• Asterisk AudioSocket
• FreeSWITCH mod_audio_stream

Starter templates

If you want a ready-to-run base for your own project, check the deepgram-starters org — there are starter repos for every language and every Deepgram product.