asterisk-ai-voice-agent

Put an AI agent on the phone. Asterisk bridges a live call to this sidecar over AudioSocket, and the sidecar runs a streaming speech-to-text → LLM → text-to-speech loop, so the caller has an actual back-and-forth conversation — interruptions and all. It's self-hosted: your PBX, your API keys, your prompts, no per-minute SaaS in the middle.

  ┌────────┐   RTP    ┌──────────┐  AudioSocket (TCP)  ┌───────────────────────┐
  │ Caller │◀───────▶│ Asterisk │◀───────────────────▶│  ai-voice-agent       │
  └────────┘          └──────────┘   slin16 8 kHz      │  STT → LLM → TTS loop │
                                                        │  + tool calling       │
                                                        └───────────┬───────────┘
                                                  Whisper/Scribe · Claude · Piper/ElevenLabs

What you get

• Speech in via OpenAI Whisper or ElevenLabs Scribe, with WebRTC VAD deciding when you've stopped talking.
• The brain is Anthropic Claude, streamed token-by-token so the agent starts replying before the whole answer is ready. Swapping in another LLM means implementing one small module interface, documented in PORTING.md.
• Speech out via Piper, which runs locally and costs nothing, or ElevenLabs if you want their voices. Either way it's resampled to the 8 kHz slin16 that Asterisk expects.
• Barge-in — start talking and the agent shuts up, like a real conversation.
• Tool calling — let the model transfer the call, schedule a callback, look something up in your CRM. Calls go out to a webhook you control, so the actual logic stays in your stack.
• Personas are just YAML: a greeting, a system prompt, which voice, which model, which tools.
• The pacing is handled. This is the part everyone gets wrong the first time (more below).
• Runs in Docker. docker compose up, point Asterisk at it, done.

How it works

1. Your dialplan answers a call and runs AudioSocket(<uuid>,<host>:9092), passing a persona name via a channel variable.
2. The sidecar accepts the AudioSocket connection, reads the UUID frame, and loads that persona from personas.yaml.
3. It speaks the greeting (TTS → AUDIO frames), then loops: caller audio → STT → on a final transcript, stream the LLM reply → buffer to sentence boundaries → TTS → paced AUDIO frames back.
4. If the LLM emits a tool_use, the sidecar POSTs it to your configured tools webhook, feeds the result back, and continues.
5. On hangup it tears the call down and (optionally) POSTs a transcript to your webhook.

The AudioSocket framing/pacing lives in a standalone, tested package — asterisk-audiosocket (Node/TypeScript) — and in asterisk_ai_voice_agent/audiosocket.py here (Python). Same wire protocol, pick your language.

A word on pacing. app_audiosocket shoves each AUDIO frame at the channel the moment it arrives. So if you synthesize a sentence and write it all at once, you overrun the far end's jitter buffer and the caller hears only the tail of every phrase — which is baffling until you figure out why. The sidecar meters outbound audio to the 20 ms frame clock and re-clamps the deadline every frame, so a slow TTS response can't make it burst to catch up. We learned this one the hard way in production; if you roll your own, steal this bit.

Quick start (Docker)

git clone https://github.com/ictinnovations/asterisk-ai-voice-agent
cd asterisk-ai-voice-agent
cp config.example.yaml config.yaml          # add your API keys
cp personas.example.yaml personas.yaml      # define your agent(s)
docker compose up -d

Quick start (pip)

Piper's phonemizer needs espeak-ng on the host, so install that first.

sudo apt-get install -y espeak-ng
pip install asterisk-ai-voice-agent

cp config.example.yaml config.yaml
cp personas.example.yaml personas.yaml
AI_AGENT_CONFIG=config.yaml AI_AGENT_PERSONAS=personas.yaml asterisk-ai-voice-agent

Add to your Asterisk extensions.conf:

#include "ai-voice-agent.conf"

Copy asterisk/ai-voice-agent.conf into /etc/asterisk/ and dialplan reload, then test — this rings your SIP phone and, when you answer, drops you into the demo persona:

# Replace PJSIP/1001 with your own endpoint (e.g. SIP/1001, PJSIP/myphone).
asterisk -rx 'originate PJSIP/1001 extension demo@ai-agent-test'

Answer the phone and talk to the agent. To route real traffic, point any inbound DID, queue, or extension at the bridge:

exten => _X.,1,Set(PERSONA=support)
 same => n,Goto(ai-agent-bridge,s,1)

Persona config

# personas.yaml
demo:
  greeting: "Hi! Thanks for calling. How can I help you today?"
  system_prompt: |
    You are a friendly, concise phone assistant for Acme Corp.
    Keep answers short and natural for speech. Never invent facts.
  llm_provider: anthropic
  llm_model: claude-sonnet-4-6
  llm_temperature: 0.4
  stt_provider: openai         # Whisper
  stt_language: en
  tts_provider: piper          # or elevenlabs
  tts_voice_id: en_US-amy-medium
  interrupt_enabled: true      # barge-in
  max_call_seconds: 900
  tools_enabled: [transfer, schedule_callback]   # posted to your webhook

Requirements

• Asterisk 18+ built with app_audiosocket / res_audiosocket.
• Python 3.10+ (or just Docker).
• API keys for your chosen providers. Piper TTS is fully local (no key, no cloud).

Configuration

config.yaml holds infrastructure + keys; personas.yaml holds agents. See the *.example.yaml files for the full annotated schema. Key sections:

Tool calling (webhook contract)

When the LLM calls a tool, the sidecar POSTs:

{ "session": "<uuid>", "tool": "transfer", "args": { "target": "queue:sales" } }

Your endpoint returns a JSON result, which is fed back to the LLM as the tool result. Implement transfer/CRM/scheduling however your stack does it. (In ICTContact these map to Asterisk AMI redirects, spool updates, and CRM connectors.)

Troubleshooting

Logs: set AI_AGENT_LOG=DEBUG (env or compose) for per-frame detail.

Related open source

• asterisk-audiosocket - the AudioSocket protocol layer on its own, in TypeScript. Prefer Node over Python? Build the agent in whatever language you like.
• asterisk-ami-node - Asterisk Manager Interface client, zero dependency. This is what you reach for to implement the transfer tool.
• freeswitch-esl-node - the same idea for the FreeSWITCH Event Socket.
• pbx-mcp - a Model Context Protocol server that gives AI assistants a read-only window into Asterisk and FreeSWITCH.
• ICTCore - the open source telephony framework behind our products.

Provenance & credits

This started life inside ICTContact, our commercial Voice/Fax/SMS/Email broadcasting and contact-center platform, where the same pipeline runs the AI Voice Agent and live voice-translation features. We pulled out the reusable core, cut the platform-specific parts (multi-tenancy, billing, our internal REST layer), and opened it up so you don't have to build the AudioSocket-to-LLM plumbing from scratch.

Maintained by ICT Innovations and ICT Vision, who have been shipping open source and commercial telephony since 2005. Written by Tahir Almas.

If this is useful to you, the wider stack behind it might be too:

• ICTPBX - white label multi tenant IP PBX, with a free community edition on GitHub
• ICTContact - contact center and unified communications, where this agent came from
• ICTDialer - auto and predictive dialer
• ICTFax - open source fax server

Questions about the commercial products go through the ICT Innovations support portal. Issues and pull requests about this project belong on GitHub, where everyone can read the answer.

License

MIT — © Tahir Almas / ICT Innovations, derived from ICTContact.