SIP providers

Open Cold Dialer works with any SIP provider that supports WebSocket (WSS) connections. Put these four values in frontend/.env.local.

SignalWire

1. Create an account at signalwire.com.
2. Create a SIP endpoint in your project.
3. Copy its SIP credentials.

VITE_SIP_URI=sip:your-extension@your-project.sip.signalwire.com
VITE_SIP_PASSWORD=your-password
VITE_SIP_WS_URL=wss://your-project.sip.signalwire.com
VITE_SIP_CALLER_ID=+1XXXXXXXXXX

Local numbers work immediately. Toll-free numbers need verification first.

Telnyx

1. Create an account at telnyx.com.
2. Create a SIP connection in Mission Control.
3. Buy a phone number.

VITE_SIP_URI=sip:username@sip.telnyx.com
VITE_SIP_PASSWORD=your-password
VITE_SIP_WS_URL=wss://sip.telnyx.com:8443
VITE_SIP_CALLER_ID=+1XXXXXXXXXX

Twilio

1. Create an account at twilio.com.
2. Create a SIP domain in the Console.
3. Buy a phone number.

VITE_SIP_URI=sip:your-sip-domain.pstn.twilio.com
VITE_SIP_PASSWORD=your-password
VITE_SIP_WS_URL=wss://your-sip-domain.pstn.twilio.com
VITE_SIP_CALLER_ID=+1XXXXXXXXXX

Asterisk, FreeSWITCH and other servers

VITE_SIP_URI=sip:username@your-server.com
VITE_SIP_PASSWORD=your-password
VITE_SIP_WS_URL=wss://your-server.com:8089
VITE_SIP_CALLER_ID=+1XXXXXXXXXX

Enable WebSocket transport on the server and serve it over TLS. Browsers allow WebRTC audio only on secure pages.

Testing without a provider

Leave the SIP variables empty and click Call on any lead. The app simulates the connecting, ringing and active states, so you can try the whole workflow first.

Audio

• The browser asks for microphone access on the first call.
• Use headphones to prevent echo.
• If nobody can hear you, check the input device in your browser's site settings.