Try IPTRAF or TCPDUMP. Denis.
Em Qui 14 Out 2004 19:12, Matthew Boehm escreveu: > 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 _______________________________________________ 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
