On Wed, Apr 16, 2014 at 1:35 PM, Consultor VOIP <[email protected]> wrote: > Hi ! My name is Gerald and I am working with WEBRTC and JsSIP. > > I configure my Asterisk 11.7.0 to work wit WEBRTC. > > Using a JsSIP (http://tryit.jssip.net/), the SIP extension can connect at > the Asterisk, but when we try to make a call they send a 488 response and > finish it. > > here is the part of the SIP DEBUG
We can't do much with part of your debug. You'll want to post a pastebin link to your full SIP trace, and be sure that it includes at least VERBOSE messages turned up to 5.[1] Work on WebRTC support is on-going, so you'll want to test in the very latest Asterisk version in your branch (11 or above). That means you need to be on 11.9.0-rc2[2] at this moment. [1]: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information [2]: http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-11.9.0-rc2.tar.gz -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com & http://asterisk.org -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
