28 Dec
2010
28 Dec
'10
11:17 p.m.
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
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users