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 UA- 192.168.2.11 OpenSIPS: 192.168.2.5 Asterisk (Test Box): 192.168.2.10 A slightly unrelated, do we have to "force_rport();" on all SIP traffic or only the initial INVITE? Thanks in Advance, and great product!!! Nick. _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
