[asterisk-users] SIP/RTP Nat problem, can't solute it.

[C F]

Change To canreinvite=no

On 1/6/07, Facundo Barrera - GMail <facubarrera at gmail.com> wrote:
> Dear list:
>             I have the typical one way audio problem, as far as i know
> it's a nating problem, my hosts inside my lan can call to outside
> internet hosts, but can't listen a thing, i read a lot about sip and
> rtp and protocols and the problem it seems to be with NAT, this is the
> config i put on my sip.conf file about nat:
>
> externhost=sip.server.com.ar > my server name on the internet
> localnet=192.168.5.0/255.255.0.0 > my LAN
> nat=yes
> canreinvite=yes
>
> And this are the ports i opened on my firewall script
>
> iptables -A INPUT  -p udp -m udp --dport 8766:35000 -j ACCEPT
> iptables -A INPUT  -p udp -m udp --dport 5004:5082 -j ACCEPT
>
>
> But still can't hear a thing from an outside call, any hel will be
> appreciate
>
> Thanks a lot
>
> --
> _________________________
>    Facundo Agustin Barrera
>   --------------------------------------
>      www.openlabs.com.ar
> "Let the penguins do the work"
> ---------------------------------------------
>    Buenos Aires - Argentina
> _________________________
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