Start your WebRTC desk phone
Click Connect Gateway first, then login with your Asterisk server, extension, and extension password.
Designed for call-center agents
This page keeps the Janus SIP plugin behavior, but removes the demo look and simplifies the SIP URI handling for daily operation.
AudioWebRTC to SIP
DTMFIVR keypad
Secure UINo SIP URI typing
Required fields
- Asterisk server address, for example
192.168.10.5 - Extension number, for example
1001 - Extension password/secret
Agent Login
Enter Asterisk details. SIP identity will be generated automatically.
Asterisk Server Address
Do not type
sip:; the panel adds it automatically.Extension
Password / Secret
Dial Console
Enter digits only. The final SIP URI is generated by JavaScript before dialing.
Dial Number
Generated call URI: sip:number@server