[Kamailio-Users] call routing problem

[bhrugu mehta]

Hi all,

I am new to openser.
I have register two sip user  in openser (as register server) and call
handling in asterisk.
when 1001 user do a call to 1002 nothing happen.
call rejected.
If posible give a sip.conf and extension.conf snap of this scenario.

any suggestion?
Bhrugu Mehta
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20090508/e79c72fa/attachment.htm>