[Asterisk-Dev] [Asterisk-Dev]SIP(ATA-186) configuration for Asterisk

[Santosh Prasad]

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