[Asterisk-Users] Asterisk SIP Gateway behind NATS - SIP/2.0 404 Not Found

[Pimjai Wesnarat]

Hi all,

I am running an Asterisk server behind a NAT.
I want to forward the calls from PSTN to a SIP phone (no nat and also an 
asterisk).
I set the externip and localnet in sip.conf already. I opened the ports 
in my firewall. (I changed SIP port from 5060 to 5065 and limited the 
rtp port to 12000-13000)
However, I just can't call out. I've always received SIP/2.0 404 Not Found.

My sip.conf looks somewhat like this

[general]
context=default            ; Default context for incoming calls

bindport=5065            ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0        ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes            ; Enable DNS SRV lookups on outbound calls

externip = 83.xxx.xxx.xxx    ; Address that we're going to put in 
outbound SIP messages
localnet=10.2.70.0/255.255.255.0
localnet=192.168.18.0/255.255.255.0; All RFC 1918 addresses are local 
networks

[thephone]
type=peer
host=thephonedomain.com
port=5065
username=abcd
nat=no
usereqphone = yes
;canreinvite=no



If I made a call to local SIP phone, it works fine. But to the SIP phone 
outside the NAT, it just doesn't seem to work.
I have no idea what else I should do.
Anybody could give me some suggestion??



regards,

Pim