Rob a écrit :
> Hi all,
>   
Hi
> I'm using GIZMO with my asterisk (1.4.13) box ... I've had CALL IN for a
> while and it works fine .... I just added CALL OUT ... I have no problem
> with call setup ... the called party hears me ... but I can't hear them ....
> again if the call comes INTO the server both sides work fine.
>
>   
Looks like a nat issue: do you have nat=yes and canreinvite=no in your 
sip.conf for Gizmo5?

-- 
Daniel

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