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

[Facundo Barrera - GMail]

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

-- 
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   Facundo Agustin Barrera
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     www.openlabs.com.ar
"Let the penguins do the work"
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   Buenos Aires - Argentina
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