Hi Kurt,
You are on the same boat as me except that I have been able to have two sip clients talk to each other and also able to route external calls via pstn. I would like to help you with your issue. Can you let me know if you created an entry for not authenticating calls from openser ? Also when you dial another extension from one extension do you see any thing in the asterisk cli ? Also if you could post me ur kamailio.cfg and sip.conf for asterisk, might be helpful to further troubleshoot the issue.

Thank You
Amit Nepal
Systems Administrator
Phoenix Internet

On 12/28/2010 2:07 PM, Kurt Mullen wrote:

I am on my 20^th (I’m not kidding) attempt to successfully complete this tutorial.

I have installed on Ubuntu 10.10 x64 Server. I installed Kamailio & Asterisk on the same server as in the tutorial.

I have two SIP clients registered, but they are not able to call each other.

No one answered my last two posts, so I hope someone can help this time.

Kurt Mullen
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