2013-02-19 17:10, Juan Carlos Agudelo skrev: > I don't have analog channels, this happens with SIP Trunk... > > Juan.. >
I've seen this with one of our sip-trunks. Our provider used opensips for that platform I think. If had the same account registered in two asterisk-servers and they answered the call at the same time, audio was from both asterisk-servers and the phone from pstn, like a 3-way conference. I also watched a presentation about rtp ports, mentioned on this list some time ago, explaining quite a bit on rtp ports and security: http://lists.digium.com/pipermail/asterisk-users/2013-January/277342.html http://media.ccc.de/browse/congress/2010/27c3-4193-en-having_fun_with_rtp.html So my guess is that if you get two devices using the same port, or one device that don't stop sending, you will hear that injected in your call. -- Johan Wilfer -- _____________________________________________________________________ -- 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
