Asterisk serializer and transport

api-reference/server/services/community-integrations.mdx

@@ -66,6 +66,7 @@ Serializers convert between frames and media streams, enabling real-time communi
 
 | Service                          | Repository                                 | Maintainer(s)                         |
 | -------------------------------- | ------------------------------------------ | ------------------------------------- |
+| [Asterisk](https://www.asterisk.org/) | https://github.com/NikolayShakin/pipecat-asterisk | [Nikolai Shakin](https://github.com/NikolayShakin) |
 | [AwaazAI](https://www.awaaz.ai/) | https://github.com/awaazde/pipecat-awaazai | [AwaazAI](https://github.com/awaazde) |
 | [Wavix](https://wavix.com/)      | https://github.com/wavix/pipecat-wavix     | [Wavix](https://github.com/wavix)     |
 

pipecat/telephony/asterisk-websocket.mdx

@@ -0,0 +1,83 @@
+---
+title: "Asterisk Websocket Channel"
+description: "Complete guide to Asterisk integration"
+---
+
+## Things you'll need
+
+- Asterisk (version >= 22.8.2) installed
+- Pipecat (version >= 1.1.0)
+- At least configured sip profile configured both on Asteriks and your sip-devece/softphone
+- API key for Gemini
+
+## Simplest setup
+- Install Docker
+- Clone the [project](https://github.com/NikolayShakin/pipecat-asterisk)
+- Add your API key for Germini in `.env`
+- Run container with asterisk `cd examples/pipecat_asterisk && docker-compose up -d`
+- Run pipecat app `uv run examples/pipecat_asterisk/ws_server.py`
+- Register your sip-phone on 127.0.0.1:5060, Dial 24
+
+## Custom setup with your Asterisk instance
+### Asterisk configuration
+
+```
+extensions.conf
+exten = 8,1,Dial(WebSocket/pipecat/c(slin)) ; Sampling rate 8kHz
+;; Optionally
+exten = 12,1,Dial(WebSocket/pipecat/c(slin12)) ; Sampling rate 8kHz
+...
+exten = 192,1,Dial(WebSocket/pipecat/c(slin192)) ; Sampling rate 192kHz
+```
+
+```
+websocket_client.conf
+[pipecat]
+type = websocket_client
+uri = ws://127.0.0.1:7860/ws ; Assuming pipecat running on the same host. Adjust the address if needed
+protocols = media
+connection_type = per_call_config
+connection_timeout = 100
+reconnect_interval = 50
+reconnect_attempts = 3
+tls_enabled = no
+```
+### Pipecat configuration
+Create and run your pipeline using `AsteriskWebsocketTransport` like this:
+```
+from fastapi import FastAPI, WebSocket
+from pipecat.pipeline.pipeline import Pipeline
+from pipecat.pipeline.task import PipelineTask
+from pipecat_asterisk import AsteriskWebsocketTransport
+
+app = FastAPI()
+
+@app.websocket("/ws")
+async def websocket_endpoint(websocket: WebSocket):
+    await websocket.accept()
+
+    # Initialize the Asterisk Transport
+    ws_transport = AsteriskWebsocketTransport(websocket=websocket)
+
+    # Build your Pipecat pipeline
+    pipeline = Pipeline([
+        ws_transport.input(),
+        # ... other pipeline components (VAD, LLM, TTS, etc.)
+        ws_transport.output(),
+    ])
+
+    task = PipelineTask(pipeline)
+
+    # Run the pipeline
+    # ...
+```
+`uv run your_pipeline.py`
+
+### Dial
+Configure your sip-phone to work with asterisk and dial `8` or any other configured extension if you want to test different sampling rates. 
+
+
+## Next Steps
+
+- Explore the [example](https://github.com/pipecat-ai/pipecat/tree/main/examples)
+- Learn about [Asterisk websocket channel](https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket/)