I have kamailio and asterisk running on same machine. When I make an internal call, it routes the call to other extension. However, if I stop the asterisk service then the call still routes the same way to other extension. Is there a chance that kamailio does the call routing it self. 

Also, I cannot see any sip extension registered at asterisk when I run a command "Sip show peers" neither there is any activity in Asterisk's log. 

Regards,

Amjad