please look at http://www.voip-info.org/wiki/view/Asterisk+SRTP
and try compile&run clients with srtp (linksys,grandstream,aastra, qutecom, eyebeam, ...) digium need feedback for srtp inclusion to 1.6.3.0 http://bugs.digium.com/view.php?id=5413 if you need additional info, i'm on jabber - [email protected] thanks! --------------------------------------- Marek Cervenka ======================================= _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
