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

[Gao]

http://www.voip-info.org/wiki/view/Asterisk+func+sip_header


On 2016-11-09 08:13 AM, Ethy H. Brito wrote:
> 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
>