2014-11-12 16:24 GMT-06:00 Dario Estupinan <darioestupi...@soygenial.co>: > tengo la siguiente pagina pero no se como seguir despues del punto 22 > > http://highsecurity.blogspot.com/2012/12/webrtc-and-asterisk-11-using-sipml5.html > > gracias!
You haven't described your problem and I'm relying mostly on Google translate, so I'm not sure what to tell you. There is another tutorial available at: https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5 If you have an issue specific to FreePBX, you might try asking about it on the FreePBX community forums: http://community.freepbx.org/ Thanks, -- 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
