VoIP Soft

SIP Over Asterisk

How do I get SIP endpoint to talk through Asterisk?

I have two SIP endpoints provisioned on the SIP PBX and I can see SIP registers correctly. However, the two SIP endpoints can not call one another (get a "404 Not Found" SIP error message in response to an INVITE). I want to add more SIP endpoints to the PBX but need to get this simple case working first.

I have configured the sip.conf file as follows:

1000
type=user
host=dynamic
context=default
username=1000
dtmfmode=rfc2833

1001
type=user
host=dynamic
context=default
username=1001
dtmfmode=rfc2833

The sip clients are registering in the format:

Any ideas on what is wrong? Do I need to do anything with the extensions.conf file (or is the default settings sufficient to get one sip phone to call the other?)

by voip-info.org

#link posted by JeMeans @ 8:27 PM