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