[asterisk-users] troubleshooting NAT

[Martin]

if you're using SIP then you look at SIP headers ... SDP part
from INVITE's and 200 OK to INVITE. You check what IP/port is used for RTP
and then you can try to get some packet dump with tcpdump/wireshark
and maybe configure your router
so it works.... it's the first thing to look for ...

you can also try to use the stun server ... asterisk has it built in
...never used it but saw it's there

Martin

On Tue, Oct 20, 2009 at 1:32 PM, Ott Rose <sixfourimpala at hotmail.com> wrote:
> Can anyone tell me how to troubleshoot NAT issues? We had Freepbx look at
> your install and they said we are having a NAT problem but didn'ttell me if
> it was with the asterisk conf or the Cisco ASA.
>
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