Leif: I take it that the * on the GW machine remains in the middle of the call between SIP and the Nat'ed *, correct? I also suspect that you will be using IAX over Ethernet (to avoid compression/decompression delays), correct? Great work! Regards, Uriel
-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Leif Madsen Sent: Sunday, September 28, 2003 3:42 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] NAT/SIP solution? Stig Hess wrote: > I meant where Asterisk is behing a NAT... sorry for the confusion. Hi Stig, If you are able to run * on your NAT'd box, then I have come up with a work around (thanks wasim!!!) that will allow you to run an * box behind your NAT, and still recieve and make SIP calls. I haven't got the whole thing figured out yet in terms of extensions.conf (but I am working on that today, will post on my website later) but this is basically my configuration: <remote> <--TDM400P--> <*> <--IAX--> <*> <--SIP--> <remote> |-----NAT-----||FW| So basically the * on the GW machine which is also the NAT / FW box recieves the connection from the SIP remote end, then forwards all the traffic over IAX to the NAT'd * box. I just tested it, and it works fine! Once I get some more complex extensions.conf files setup, I will post them. Thanks, Leif Madsen. BTW: As for just passing SIP through SIP, I believe it's a limitation of the SIP protocol as the RTP ports are different than the connection port, whereas IAX is all the same port for everything (from what I gather) _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
