Is it possible to configure opensips to request the a call fail on no answer and busy. I've got the asterisk <> opensips working but it only sends calls to vm if the sip phone is not registered. I've looks thru the docs and cant find the answer. my subscriber come into opensips and routed from there . do i need to send the calls thru asterisk and have asterisk monitor them?
Yes, you can configure opensips to send calls to voice mail on no answer or busy. You have to use the failure_route for this ( the failure_route is called when a request receives only negative replies on all branches. In there you can just relay the initial invite to asterisk. Read more here: http://www.opensips.org/Resources/DocsCoreRoutes#toc3 .