Using asterisk as WebRTC2Sip gateway only

we want to use Asterisk as WebRTC2SIP gateway only by using jssip on the client side and an existing PBX on the customers side.

We tried the configuration described in docs but it seems to be an example where asterisk is the PBX.

How do we have to configure Asterisk for that purpose?


What exactly do you want it to do? If you just want it to forward calls then you’d write dialplan that uses Dial() to call another endpoint.

Maybe this is also a lack of understanding.

We just want to establish communication between a web based client and a PBX.
We want to call and to receive calls etc.

The web client has all information like registrar, user, password and the number to call.
When we use the sample configuration it tries to register with asterisk instead of the registrar.

So, we want to tell asterisk just to communicate with another PBX.

Asterisk isn’t a SIP proxy, it doesn’t forward requests around and optionally do media conversion. Each call leg is independent and incoming registers are handled locally. Without knowing the full scope of the deployment and how things work for your setup, can’t really say whether Asterisk would be the right fit or not.


What you need is Kamailio/OpenSIPS + rtpengine.

Definitely not Asterisk, unless you pass registrar functions to Asterisk via realtime mechanism or so

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