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