AgentSIP connects Large Language Models to real incoming phone calls via SIP. Build voice agents that answer, talk, and act — through Asterisk, FreeSWITCH, FreePBX, or any inbound SIP trunk.
Everything you need to run AI voice agents for inbound calls in production.
Low-latency streaming STT → LLM → TTS designed for natural, interruptible conversations.
Register as a SIP endpoint and receive calls from your PBX / inbound SIP trunk. No proprietary system required.
Swap OpenAI, Deepgram, ElevenLabs, Whisper, and more. Bring your own keys and models.
Incoming SIP/RTP audio is streamed in real-time through your chosen AI providers and back to the caller.
git clone https://github.com/fitraditya/agentsip.git cd agentsip cp .env.example .env # add your API keys docker compose up # AgentSIP is now ready to receive inbound SIP calls!
*Update this snippet to match the exact setup from your README.
Answer inbound calls 24/7, route callers, take messages, and answer common questions automatically.
Replace "Press 1 for sales" with natural conversation that understands caller intent.
Let callers book, reschedule, or cancel appointments, and get information — just by calling in.
Star the project, try it with your own PBX, and contribute.
Star AgentSIP on GitHub