I don't think the problem is NAT-related. Looks like "To" header
in SIP
INVITE message do not match to "User ID" in sipura settings.
On Thu, 2005-10-27 at 01:57 +0330, mohammad mirzaee
wrote:> Hi ALL;
>
>
> I have users with Sipura/Linksys phones regsitered behind Nat(
> useing STUN at phone not portforwarding ) in my Asterisk box, when I
> try to call them with another phone i got:
>
> Got SIP response 404 "Not Found" back from 217.6.190.4
> SIP/217.6.190.4:5060-666d is circuit-busy
>
> Is above mentioned problem relates to "Nat", Is there anybody
who use
> sipura with STUN method and can recive calls?
>
>
> My asterisk Sip.conf for Nat has the following:
>
> [sipura]
> ..
>
>
> nat=yes
> canreinvite=no
> qualify=1000
>
>
> Appreciate any help
> Mohammad
>
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