Displaying 1 result from an estimated 1 matches for "asteriskgary".
2014 May 21
1
One Way Audio with WebRTC (with external asterisk)
Hi,
I've run into a slight issue when using WebRTC and two Asterisk boxes.
I am using SIPml as the test WebRTC client.
My two asterisk boxes, one of them is configured for WebRTC with websockets, etc (asteriskrtc.local) and the other is just a standard asterisk server (asteriskgary.local).
Dealing with just the WebRTC asterisk server, asteriskrtc.local, I am able to log in to the SIPml webpage and make a call from a SIP Phone to that WebRTC user, and vice versa, and all the media flows.
When I try making a call from the other asterisk server (asteriskgary.local) to asterisk...