[Asterisk-Users] Callback application

[Adam Goryachev]

On Mon, 2005-04-11 at 16:40 +1000, Rod Bacon wrote:
> I don't know if what you're trying to do is possible, but the easiest way to 
> check would be to take a look at the raw packets on the ethernet interface 
> of your * server once a call is in progress. If indeed the RTP can be handed 
> off to the 2 endpoints, you should only see SIP traffic at your server. 
> TCPDUMP is your friend.

or sip debug, or iptraf/jnettop/any other network traffic monitor.

Regards,
Adam

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Adam Goryachev
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