asterisk users - Feb 2010 - How to tell if asterisk is handling media or not?

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!

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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On Fri, 26 Feb 2010, Alejandro Recarey 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?
Running

   iftop

on the box in a terminal window makes it relatively easy to see what's 
going where.

There are lots of things in asterisk that'll stop this working though - 
mostly detailled here:

   http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite

Good luck!

Gordon