Hi, I have a question regarding NAT - I have two Asterisk setups, and a couple of softphones on my laptop to test them. In the first Asterisk I've got nat=yes for all SIP phones. The second setup is identical as far as software is concerned, but the server is running on a VPS with one of the larger VPS hosting services.
On this second setup I was able to phone out from my XLite softphone but when I tried phoning in nothing happened, basically because the phone was always UNREACHABLE. It would register fine with Asterisk but then disappear. After nosing around for quite a bit I found a suggestion that I try setting nat=route for the SIP phone, and suddenly it worked both ways. Another SIP phone on the same laptop connects to the first Asterisk server setup (which runs on a dedicated box with fixed IP, not in VPS) but I notice that every minute or so Asterisk tells me that the phone is unreachable, then a few seconds later it becomes reachable again. My laptop is currently sitting at home with typical home-Internet configuration (ADSL, Nat, no fixed IP). I did see something somewhere about the big VPS providers using some form of "hidden" NAT but I don't know what that could mean. My question is: Does this difference in behaviour have something to do with the second server running on a VPS - and are there any drawbacks to using nat=route on all client SIP phones? Best regards Binni Care Solutions Brettesvillesgade 14 9000 Aalborg Telefon: 8832 1600 Mobil: 3020 0868 www.netklinik.dk, www.gangweb.dk, www.caresolutions.dk -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
