> > > I still think we need a SIP_CAUSE channel variable. :-) > Then we need to start working on aggregation rules, like what if one IAX channel answers and one SIP channel is busy?
For SIP-only calls, we need to add a lot of code from proxy rules for call forking and response aggregation. It's not an easy task. /O _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
