Thank you so much for your response. I should have mentioned that I changed IPs, phone numbers, and domain names to bogus values. Never know who's reading, and I would not do that to anyone either.
Irregardless, it turned out that setting nat=yes in sip.conf solved the problem. Cheers, Nick. On 1/6/13, dotnetdub <[email protected]> wrote: > On 6 January 2013 20:39, Nick Khamis <[email protected]> wrote: >> Hello Everyone, >> >> After getting out Asterisk machines up and running, two way audio >> etc... We would like to put an OpenSIPS server between the world and >> our Asterisk boxes as one would imagine. The initial INVITE is getting >> routed correctly however, the "Giving a try" from our SIP trunk is not >> making it's way to the Asterisk box. >> >> I have attached some pastebins of: >> >> Asterisk SIP Log: http://pastebin.com/VdtAKBH9 >> OpenSIPS Log: http://pastebin.com/BLcgFrV5 >> OpenSIPS Debug (Console): http://pastebin.com/gN6hgsxz >> > > > I don't think we are seeing the full picture here... What is 108.59.7.123 ? > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
