[Asterisk-Users] SIP + NAT = horrible mess

[Nabeel Jafferali]

> I don't think you can use NAT = yes unless there is a STUN
> server involved.  See my post yesterday for my Grandstream settings.

No, I had nat=yes working with my Cisco 7960 which did not provide it's
public IP. However, you need to tell the IP Phone to start using the IP
and port that * received the SIP messages from for RTP traffic (use via
IP address and via port).

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Nabeel Jafferali
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