I'm new to all this so I don't know where to look, some tips would be most appreciated. I've enabled sip debugging and everything looks fine on the client and server side. Using Linphone on the client side.
GSM playback from the server console is fine. I've used Linphone to connect to a vegastream VoIP system so I know if that installed and working. I'm basically just trying to get the sample configs working, dialing in to [EMAIL PROTECTED] It connects and playback of the demo ensues, but at the client end it's unrecognisable garbage. Any hints ? -- Glen Gray <[EMAIL PROTECTED]> 17 Dame Court Senior Software Engineer Dublin 2, Ireland Lincor Solutions Ltd. Ph: +353 (0) 1 6746413 _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
