Incomming VoIP calls problem - no sound

I have an Asterisk setup at my office that is handling outgoing and local calls perfectly. The problem is when I try to call a number of the asterisk inner network from outside. The phone rings, but there is no sound. I have my firewall setup according to this instructions given here:

[quote]# SIP on UDP port 5060. Other SIP servers may need TCP port 5060 as well
iptables -A INPUT -p udp -m udp --dport 5060 -j ACCEPT

IAX2- the IAX protocol

iptables -A INPUT -p udp -m udp --dport 4569 -j ACCEPT

IAX - most have switched to IAX v2, or ought to

iptables -A INPUT -p udp -m udp --dport 5036 -j ACCEPT

RTP - the media stream

iptables -A INPUT -p udp -m udp --dport 10000:20000 -j ACCEPT

MGCP - if you use media gateway control protocol in your configuration

iptables -A INPUT -p udp -m udp --dport 2727 -j ACCEPT [/quote] … wall+rules
I have also added the Asterisk server to the DMZ.

The firewall/router is a DrayTek Vigor 2910 with 3.2.1 firmware.

I use a Lynksis SPA2102 to make the test calls. The SPA2102 goes out through a different provider, but it is in the same network as the Asterisk server. When I try calling the server I use the following address. 122@domain_of_my_network. I don’t hear a sound when I pick up the phone, but it rings. When I call 122@local_ip_of_the_Asterisk there is no problem and I can hear perfectly. What am I doing wrong? From what I know it should be a problem with the RTP ports, but I have them configured as the should be, right?

This is a funny set up dude… Am i correct in saying you are trying to ring your server which is local to the calling device, through an VOIP provider located externally?

Becasue of the way you have set this up, there are about 5 possible causes to this problem, most likely a mixture of an issue with the NAT on your router and a reinvite request directing the packets incorrectly. Can you post the SIP debug output from Asterisk? Also log packet drops on your firewall and see if anything is getting dropped there.

A more robust approach would be to set up the SPA2102 to register to your asterisk box (rather than the VOIP) and set the VOIP up as a trunk on asterisk.