[Asterisk-Users] Asterisk+Nat+sipura (Help)

[Sergey Okhapkin]

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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