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.

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Editions

	Community	Commerce
License	MIT	Commercial
SIP Proxy + Media	✅	✅
Queue / ACD	✅	✅
HTTP Router (dynamic routing)	✅	✅
Call Recording + SipFlow	✅	✅
Transcript (SenseVoice offline)	✅	✅
Web Console	✅	✅
Voip Wholesale (VOS3000 alternative)	❌	✅
Voicemail Pro	❌	✅
Enterprise Auth (LDAP/SAML/MFA)	❌	✅
Endpoint Manager (phone auto-provisioning)	❌	✅

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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

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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

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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

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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.

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Screenshots

Extensions	Call Records	Route Editor

Transcript	SIP Flow	WebRTC Phone

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Documentation

Configuration Guide	All config options
API Integration Guide	HTTP Router, Webhooks, Active Call Control
Product Roadmap	Commercial plugins & Q2 2026 plan

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Troubleshooting

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

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

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License

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