Hi I am working on setting up Asterisk-openH.323-SIP. I am able to setup Asterisk and openh323. I am using ATA 186 to be my SIP endpoint. I am not sure if my configuration is correct, I have followed the steps given in the following website: http://www.djernes.org/~shawn/ata186.htm I am able to dial the demo extension 1000 and speak to a Digium representative from the SIP phone. But I am unable to talk between the other SIP/H.323 endpoints in my network.
When I do sip debug I get the following error: Looking for 4010 in default Transmitting (no NAT): SIP/2.0 404 Not Found Can anyone suggest me some ideas of where I have to look for correcting these errors. Best Santosh _______________________________________________ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev
