About NAT: According to some info on the net forwarding should not be required. I think that the real and true answer in that case is that forwarding will be required some timens and some times not, depending on the server technology of the ip telephone vendor. I have tried different clients and also to dump the trafic with ethereal, and I think they do not do it the same way, allways.
I have made one installation of Asterix@Home at my SME 7.0 Gateway and one other at a PC on lan. To day I have finally managed to hack my adsl router, so it is not a router any more but a bridge. The SME gateway now receive the external ip.
I think it is right to say that a central part of the functionality of the Astersis server is to work as a sip telephone proxy server and a a local sip telephone switcboard. In this way it has the nature of a client releative to the external sip server you are connecting to and as a server for your local sip telephone. In this way it lookes like it can "translate technology standards" so you can have one internal protocol standard in your home/office while different oubound lines (trunks) can run different protocol standards (Also at Asterisk@home, or will it be needed "Asterisk normal" ??) while your clients have only one communication setup.
I think that if you have only one ip telephone you will have rather little use for a Asterisk proxy/telephone switchboard. You can connect your telephone client directely to the external server or via the Asterisk proxy/switchboard. There should not be the big differences. (exept for some increased functionalities of the Arsteisk like telephone ques, music while you are waitning, anwsering machine etc.)
One other big and important reason will of cource be if you want to learn the new technology.
Because of "the client nature" of the Asterisk "proxy server" I believe that the case will be that if you will have to make a forwarding to make a client to work you will also have to do the same if you are running a Asterisk server at lan or a kind of dmz.
On the other hand, if you are running the Asterisk on a gateway, it might be situations where you also will have to open aditional ports on the gateway to make it work.
I use the norweigan sip telephone wendors Telio and Televoip but I have not come so far, yet that I have been able to connect to their sip servers via Asterisk.
I have until now just tried with some free services like pulver / freeworlddialup. This does not work for me so far.
I just try to explain things in such a way that I am able to understand it myself
Please correct me if I should be wrong.
By the way, I think the SME version of the Asterisk@home and the orginal downloadable "Centosbundle" work rather the same way. Initially revision status seems to be approx the same, but the "Centosbundle" did a lot of updating using the command "yum uptdate".
Hopefully "someone" will correct me if I'm wrong about "technology and understanding". (For me one or two ip telephone clients will be enough so the Astersik server part of it is mainly for learning - and learning that will normally be to do some mistakes and inncorrect conclusions on the way.