I have installed the SAIL contribution (I think it is SAIL - lots of things seem to be installed, each with brand names, leaving me a little confused as to what part each plays in the overall 'Asterisk' server) on SME Server 7.3, and am trying to figure out how it all hangs together. One thing I would like to do is to be able to accept incoming SIP connections direct from SIP phones.
For example, I would like to dial "sip:5000@example.com" or "sip:5000@12.34.56.78" into the xtem softphone, and have it dial through to extension 5000 on my server at that domain or IP address.
Is that possible? If so, can it be done through the SAIL admin panels, or would it require some hacking? I am assuming that using 'sip:' in a softphone means that phone will attempt to go *direct* to that PBX - is that right, or have I got the wrong end of the stick?
If I do this, is it a potential security problem, or a gaping hole allowing voice spammers to dial all my extensions in bulk?
-- Jason