WebRTC custom trunk call


I have a full configured sipML5 webrtc, and my asterisk serveur with one Trunk = Trunk A

Is it possible to do a custom call throught an other trunk.

As an example :
A user want to call someone trought my webRTC but with his Trunk B not my Trunk A. So the user tell to the webRTC that he want to call with his Trunk B by inserting all informations of his trunk we need (password, domain, username…) (informations receive with sip-header)

I want to custom my dialplan to check if the user using a custom webrtc or not.
If it’s true => then call with his Trunk B (NOT the Trunk A)
If it’s false/empty => then call with Trunk A

Any advices will help !

Is it possible to call ${SIP_HEADER(password)} in peer details to get trunk info from sip_headers??

I tested it, but i have a failed auth, how do i call sip_header variables in sip.conf ??

I wan’t to call Sip_hearder variables in the sip.conf to have a dynamic trunk, is it possible that way?


Is it something I should not ask? There is no way to have a dynamic trunk?

Not with asterisk. You could set up a sip proxy to do what you want, maybe.

thought that was a basic feature… i really need this feature up. But there not so much docs on google, do you have any way to achieve that? like a forum / tuto links

Thanks for your reply

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