I have a block of 20 IP addresses, so I can't really carve out /30s and whatnot to route between locations.
Asterisk is the client. I am doing Interop testing with some vendors before I ship it out to a colo facility. I have used the NAT setting with Asterisk as the server on the open Internet. Would it function similarly with Asterisk as the client? We are using IP based authentication. I have more public addresses, but I'm unsure how to route that through so the Asterisk box can use it. Perhaps I will look into 1:1 NAT. --Mike _____ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez Sent: Thursday, March 29, 2007 11:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] SIP & NAT What do you mean by 'non-standard' IP block? Is the Asterisk machine behind a NAT, or are only your clients? Did you look at the nat setting sin sip.conf? Do you have a static public address that can be routed to the Asterisk box? _____ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Hammett Sent: Thursday, March 29, 2007 11:52 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] SIP & NAT I hate SIP. The only reason I'm doing this is that its cheaper than deploying the server to a colo facility. My provider has given me a non-standard IP block, so I can't do typical routing. I have Asterisk server <-> MT w\NAT <-> PPPoE <-> MT <-> Provider. I setup a dst-nat on 5060 to the Asterisk box. Audio from Asterisk --> PSTN works great. Audio Asterisk <-- PSTN does not. Ideas?
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