On Fri, 2007-03-23 at 16:12 +0100, marek cervenka wrote: > please look at > http://www.voip-info.org/wiki/view/Asterisk+SRTP > > and try compile&run clients with srtp (linksys,gxp-2000, minisip, twikle, > ...)
Does this work on 1.2 or 1.4 too or is it trunk only? Regards, Patrick _______________________________________________ --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
