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

[C F]

In 1.2 you can use rtp debug in the CLI

On Thu, Feb 25, 2010 at 8:27 PM, Alejandro Recarey
<alexrecarey at gmail.com> wrote:
> 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!
>
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