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SIP / baresip Bridge

Turn your hive into something you can call on the phone. The HiveMind-baresip-bridge answers ordinary SIP calls with baresip, a SIP softphone, and turns the call into a voice conversation with your hive. There's no wakeword — an answered call is the activation signal — and the caller's replies are spoken back into the call.

In a nutshell

  • Runs the hivemind-baresip-bridge console command, wrapping a local baresip process.
  • No wakeword: an answered call is the activation. Local VAD segments the caller's speech.
  • The bridge asks hivemind-core to synthesize replies (speak:b64_audio) and plays the audio into the call — it does not synthesize speech itself.

Beginner's mental model

A phone call to your hive works like a voice satellite, except the microphone is a phone call instead of a room. Whoever calls in gets to talk to your assistant.

Getting a SIP account

You need something that can route a call to baresip:

  • Self-host Asterisk, a SIP server, and register two extensions on it — one for the bridge, one for whatever device calls in. Asterisk only needs to be reachable by baresip and the calling device, so a LAN or a Tailscale network is enough; it doesn't need to be public. The extension number and password become the bridge's sip_user / sip_password, and the Asterisk host becomes sip_gateway.
  • Use a SIP trunk provider. Any VoIP reseller selling SIP trunking gives you a username, password, and gateway hostname — the same three values, at the cost of a subscription, but with a real phone number the public can dial.

Install

Not on PyPI yet

pip install HiveMind-baresip-bridge fails (issue #5). Install from source, or build the repo's Dockerfile, which also installs a working baresip binary.

git clone https://github.com/JarbasHiveMind/HiveMind-baresip-bridge
cd HiveMind-baresip-bridge
pip install .

A working baresip binary must be on PATH (apt install baresip on Debian/Ubuntu), or use docker build -t hivemind-baresip-bridge . instead.

Configuration

SIP settings live in a JSON file, ~/.hivemind_baresip_bridge.json by default:

{
  "sip_user": "1000",
  "sip_password": "secret",
  "sip_gateway": "sip.example.com",
  "sip_transport": "udp",
  "auto_answer": true,
  "allowlist": []
}

allowlist restricts which caller numbers get answered; leave it empty to accept any caller. HiveMind hub credentials (access key, password, host) are set separately, the same way as any other HiveMind client.

Permissions

hivemind-core add-client --name baresip-bridge
hivemind-core allow-msg "recognizer_loop:utterance" <id>
hivemind-core allow-msg "speak:b64_audio" <id>
hivemind-core allow-msg "baresip.dtmf" <id>

Skipping speak:b64_audio is the classic failure mode: calls connect and get transcribed fine, but the bridge never speaks back, because the hub silently drops the reply-audio message instead of erroring.

Run

hivemind-baresip-bridge --host ws://core.example.com --key <access-key> --password <password>

Call the SIP extension you configured. You should hear the assistant's reply spoken back into the call.

Source

Validated against the HiveMind source:

  • README.md — architecture, install, configuration
  • docs/setup.md — full Asterisk walkthrough and troubleshooting