[asterisk-users] How to tell if asterisk is handling media or not?

[Alejandro Recarey]

I'm trying to get my asterisk server to reinvite. I have two asterisk
servers with public IP's. My users (behind NAT) register on one server
(I'll call it server 1), and for some calls they are transfered over
to the other server (server 2), because that server has the E1's.

I want server 1 to be in the signaling path for billing reasons, but
handling the media stream is killing my capacity, and it should not be
necessary as server 2 also has a public IP address.

I have tried playing around with the "canreinvite" options in sip.conf
but the problem is I cannot tell if asterisk is reinviting the call or
not.

How can I figure out where the media stream is going?

thanks!