On Wed, Feb 29, 2012 at 10:34 AM, Steve Totaro <stot...@asteriskhelpdesk.com> wrote: > >
[...] > If you can post some SIP debug info from an ATA trying to register without > any redirection and also the relevant portions of your sip.conf, I am sure I > can help. > > Do it from a new location with an el cheapo home router, Linksys WRTXXX. > Yeah, I think it's time for me to shut up about SIP/NAT problems and, like you Carlos and Kevin pointed out, run a clean un-contaminated test lab to see if we can determine why our current set-up is so problematic with SIP and NAT. > If I cannot help you in a few emails, we can take this offline. > Thanks for offering to help. I will set-up a test lab but it's gonna take me some time to free a public server to do so. But it is obvious that the problem is on our side after reading all the responses. After all, VoIP is *not* by any means our core business we just use it as a tool, and up until now I thought that *everyone* using SIP ATAs and Asterisk had these NAT woes, so we just assumed it was so, and thought that mostly everyone had to perform particular configurations on the endpoints. It now seems obvious we are wrong. Anyway, my whole argumentative line in this thread is that in our particular case we found that IAX2 works great for _our_ set-ups and we don't share the view that IAX2 is a broken bat, and that in fact for us it just works great. Thanks, -- Alejandro Imass -- _____________________________________________________________________ -- 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