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Originally Posted by MarkosJal
I think your first mistake is putting the computer in the DMX , this rffectively sends all inbounds to your computer.
If you MUST have your computer in the DMZ then forward both the SIP and RTP ports to the Telephone adapter
In a Sipura look for RTP Min and RTP max those are the RTP Ports, on the SIP Page usually 16384 to 16???. The SIP Ports are on the Line 1 and line 2 Pages usually 5060 and 5061. If you do this correctly toy will not need STUN or any other solution.
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I've done the following:
UDP 5060-5070 to ATA ip
UDP 8766-35000 to ATA ip
Calls still go straight to mail. How can I track the problem, is there a "debug mode" in the website?
PS: when directly registered to voipdiscount I receive the Sip Invite from softphone