[Kamailio-Users] call routing problem

[Iñaki Baz Castillo]

2009/5/8 bhrugu mehta <bhrugusmehta at gmail.com>:
> 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?

Do a SIP capture with ngrep to understand the problem.

-- 
Iñaki Baz Castillo
<ibc at aliax.net>