[asterisk-users] Behind NAT: source is fring software (SIP)

[bilal ghayyad]

Hi All;

My Asterisk is behind NAT with IP Address 192.168.0.2. I configued on my iPlanet router and port forwarding for 5060 (UDP) to be forwarded for 192.168.0.2 and I was able to let the fring softphone (SIP) to register on the asterisk.

But when caller initiate call, the caller hear the destination but the destination does not hear the caller.

I checked the RTP port range and I found it (10000 - 20000) and I forwarded it for the internal IP address 192.168.0.2 but the problem stayed!!

I do not know what should I do more? What it could be the reason for the problem? What should I do on the router more?

I am also thinking if the fring software could use UDP ports other than the range setted in the rtp.conf? Is it possible that source to use different port than the Asterisk RTP ports?

Note: do I have to do port forwarding on my router for the RTP UDP ports, or it is enough to forward the 5060 UDP port for the internal IP address 192.168.0.2?

Any help?

Regards
Bilal