I'm not running X or any kind of GTK/GUI abilities on our asterisk server. I need some sort of ability to test wether sip canreinvite is working.
If it is, then the network usage should be minimal/nonexistant because all voice packets should be going phone-to-phone. If it is not, then network usage would be high because all voice packets would be going phone-to-asterisk-to-phone Does anyone know of a nice ncurses or terminal based realtime network usage app? Or is there some other way in asterisk I can tell if the phones are talking to each other directly? Thanks, Matthew _______________________________________________ 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
