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bg604

New Around Here
I just set up a VOIP and I get a dial tone but when I dial out to any number I get a fast busy signal or when I call my number it says its not available.

What are some things I can try to change on the router?
firewall, dmz, other settings?
 
First assign the VOIP device a static IP then put that IP into the DMZ.

If that fixes the problem contact the VOIP hardware manufacture and find out what ports and protocols need to be forwarded.

It may not be that risky to leave the VOIP in the DMZ but since you can only put one device into the DMZ and you might need that functionality for some other situation in the future port forwarding is a more elegant solution.
 
There is no need to use DMZ. Just give your ATA a static IP, and see your provider documentation on the recommended settings. I use a Cisco SPA112 myself with voip.ms, works fine. No DMZ, no port forwarding, the only configuration I did on the router was to reserve a static IP.

If your provider can't help you with the ATA you are using then check VoIP.ms's own Wiki. They document a lot of devices, and since they use Asterisk, there's a good chance that their settings would work for you as well.

Make sure you did NOT disable the SIP helper option on the NAT page of the router.
 
^ thanks for your help

I have a HandyTone 502 (HT502)

From the website
"The HT502 is a powerful VoIP router. The product's inclusion of an integrated high performance NAT router and dual 10/100Mbps Ethernet WAN and LAN ports enables a shared broadband connection between multiple ethernet devices. In addition to being SIP 2.0 standard compliant, the product supports Universal Plug-in-Play (UPnP), up to 2 SIP account profiles, and advanced telephony features."
 
^ thanks for your help

I have a HandyTone 502 (HT502)

From the website
"The HT502 is a powerful VoIP router. The product's inclusion of an integrated high performance NAT router and dual 10/100Mbps Ethernet WAN and LAN ports enables a shared broadband connection between multiple ethernet devices. In addition to being SIP 2.0 standard compliant, the product supports Universal Plug-in-Play (UPnP), up to 2 SIP account profiles, and advanced telephony features."

There's your problem - that device acts as a router, leading to a double NAT situation. See if that device can be configured as a pure ATA device instead of a router+ATA combo, so all the routing is done by the RT-N66U.
 

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