RustPBX

A high-performance Software-Defined PBX built in Rust. Designed as a programmable foundation for AI-powered call centers — route, record, transcribe, and webhook everything.

Different from Asterisk/FreeSWITCH (C modules), RustPBX exposes all call control via HTTP/Webhook, so any language can drive call logic.

Note: The Voice Agent functionality has been moved to a separate repository: Active Call. This repository now focuses on the SIP Proxy and PBX features.

Editions

Core Capabilities

• SIP Proxy — Full SIP stack (UDP/TCP/WebSocket), registration, auth, B2BUA
• HTTP Router — Every INVITE hits your webhook; you return routing decision in JSON
• Queue / ACD — Sequential or parallel agent ringing, hold music, fallback actions
• Media Proxy — RTP relay, NAT traversal, WebRTC ↔ SIP bridging
• SipFlow Recording — Unified SIP+RTP capture; date-organized, query-on-demand (no file-handle exhaustion)
• Transcript — Post-call transcription via local SenseVoice (offline, no cloud dependency)
• CDR Webhooks — Push call detail records + recordings to your system on hangup
• WebRTC Phone — Built-in browser softphone for testing

Quick Start (Docker)

# Commerce image (includes Wholesale + all commercial plugins)
docker pull docker.cnb.cool/miuda.ai/rustpbx:latest

# Community image
docker pull ghcr.io/restsend/rustpbx:latest

Minimal config.toml:

http_addr = "0.0.0.0:8080"
database_url = "sqlite://rustpbx.sqlite3"

[console]
base_path = "/console"
allow_registration = false

[proxy]
addr = "0.0.0.0"
udp_port = 5060
modules = ["auth", "registrar", "call"]

[[proxy.user_backends]]
type = "memory"
users = [{ username = "1001", password = "password" }]

[sipflow]
type = "local"
root = "./config/cdr"
subdirs = "hourly"

docker run -d --name rustpbx --net host \
  -v $(pwd)/config.toml:/app/config.toml \
  -v $(pwd)/config:/app/config \
  ghcr.io/restsend/rustpbx:latest --conf /app/config.toml

# Create first admin
docker exec rustpbx /app/rustpbx --conf /app/config.toml \
  --super-username admin --super-password changeme

Web console: http://localhost:8080/console/
SIP proxy: udp://localhost:5060

Build from Source

Dependencies (Linux):

apt-get install -y cmake pkg-config libasound2-dev libssl-dev libopus-dev

macOS:

brew install cmake openssl pkg-config

git clone https://github.com/restsend/rustpbx
cd rustpbx
cargo build --release
cargo run --bin rustpbx -- --conf config.toml.example

Cross-compilation for aarch64 / x86_64 via cross:

cargo install cross
cross build --release --target aarch64-unknown-linux-gnu

HTTP Router — The Key Extension Point

RustPBX calls your API on every incoming INVITE. You decide what happens:

[proxy.http_router]
url = "https://your-api.com/route"
timeout_ms = 3000

// POST to your endpoint:
{ "call_id": "abc-123", "from": "sip:+861390000@trunk", "to": "sip:400800", "direction": "inbound" }

// Your response:
{ "action": "forward", "targets": ["sip:ai-agent@internal"], "record": true }

Actions: forward · reject · abort · spam
See API Integration Guide for the full webhook and active call control reference.

Screenshots

Extensions	Call Records	Route Editor

Transcript	SIP Flow	WebRTC Phone

Documentation

Troubleshooting

SIP 401 behind NAT/Docker — set the realm explicitly:

[proxy]
realms = ["your-public-ip:5060"]

License

Community edition: MIT
Commercial edition : hi@miuda.ai