[SR-Users] Kamailio 3.1.x and Asterisk 1.6.2 Realtime Integration using Asterisk Database

[Amit Nepal]

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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