[asterisk-users] SIP and RTP port and IP addresses

[Ethy H. Brito]

Hi all

I'd like to log the client IP addr and port used for SIP and RTP *during* in a
call.

The IPs must be the real source IPs (internet accessible).

How are these parameters available from dialplan?

For instance, ${SIPURI} holds the internal "IP:port" if the client is behind NAT. 
I need the external IP:port

Regards

Ethy