Remember clients send packets from a random high port number which changes. Port forwarding on your router is pretty useless. nat=yes combined with qualify=yes should cause enough traffic on the right ports to keep the NAT translations open on your NAT router.
Brian McCrary wrote:
In theory, it sounds as if should work pretty easily, since there is a connection already established between the ATA and Asterisk. One big think I forgot to mention is my NAT device is a MC3810 with a PRI attached to it, so it already is running it's own SIP user agent. So, when the ATA gets a call, Asterisk forwards it to the MC3810 (since it's IP registered with the ATA through NAT.) Since the MC3810 doesn't know about the ATAs it just returns "404 Not Found", and that's the end of it. So, I'm guessing, perhaps having a SIP user agent running on the NAT router itself could be causing some of the problem? It's frustrating to see Asterisk talking to the phone with OPTIONS packets, but then not be able to send a call to it, seems like I must be close.
Everything I said assumes that the NAT router is NOT "SIP Aware". On the Ciscos I manage I turn off their "SIP fixup" feature. Mostly because I want them to act like Linksys, or whatever, that our users have at home.
If your NAT router is "SIP Aware", then you don't need portforwarding, qualify=yes or nat=yes. You should have nat=no because the NAT router is doing all the magic and Asterisk's NAT magic may not work with the router's NAT magic.
Search the archives for "MC3810" or other keywords relaying to the router. I seem to recall someone else mentioning the same model.
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