-- Auto fallthrough, channel 'SIP/xxxxxxx-00000024' status is 'CHANUNAVAIL'

The incoming call shows no WebRTC. If it’s WebRTC, then whatever you are connecting through is dealing with the WebRTC and making it non-WebRTC from the perspective of Asterisk.

My office Call Center is running with Vicidial and asterisk How they configured to call from browser?Sip over Websocket?Or is there something?

If you are connecting to such a thing, it is probably not over WebRTC from the Asterisk you are trying and instead just normal SIP.

So i need to use Sip Over Websocket to achieve what i need?

You aren’t using WebRTC between this Asterisk and Vicidial. It appears to be normal SIP. Don’t configure anything in this Asterisk for WebRTC.

I’m also going to bow out of this thread for now, someone else may chime in.

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Thank you have a nice day.

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