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

Santosh Prasad sprasad at hubris.net
Fri Jun 20 09:04:10 MST 2003


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:
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.



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