[Asterisk-Users] how can I test canreinvite effectivness?

[Matthew Boehm]

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