NAT Problem


I am trying to configure SIP on asterisk server. My SIP server is behing the Router. The IP address of SIP server is I have forwardered UDP port 5060 from the router onto I can call and listen to the other people using xten soft phone when I am in the internal LAN i.e.

My Problem in this case is if someone else calls me from the outside/internet, then the xten phone rings. I can call outside and anyone can call me inside. But, I cannot hear the voice. There is no voice at all.

I have forwarded RTP ports to the asterisk server as well. I belive RTP port that asterisk by default uses are 10000-20000 and its mentinoed in rtp.conf.

In sip.conf I have put the following NAT related entries.

externip = mypublicip
localnet =
nat = yes

Can anyone please guide me in this record what I am doing wrong here.

Thanks in Advance.



Maybe this link can be useful for you:

Please post your entire sip.conf. Something is missing.

Open ports (all UDP !!):


No harm in the latter but some phones are using RTP ports outside the 10000-20000 range.

You might want to check this line in your sip.conf

depending on your network setup it might be needed.

Make sure the correspondending SIP blocks for the phones are correct, esp. the “NAT=YES” and canreinvite lines.

RichardHH, mirceahuh and David,

Thank you all of you replying to my post. I have already resolved the problem. The problem was with the port forwarding on the router. I was doing that wrongly. Fixed that and every thing is crystal clear now.

Thanks again