[Kamailio-Users] call routing problem

[Henning Westerholt]

On Freitag, 8. Mai 2009, bhrugu mehta wrote:
> 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?

Hi Bhrugu,

you could also add some "xlog" statements (take a look to the xlog module 
documentation how to use this function) to your configuration, and then take a 
look to the log file during call routing to get an idea how the message is 
routed and finally rejected.

Cheers,

Henning
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