Two * servers: *a and *b.
Outside call comes in *b, and is automatically routed to *a. Someone on a sip phone connected to *a then decides to transfer the call to someone on a sip phone connected to *b. The transfer works.
At this point, is *a still in the converstation? Or is * smart enough to see where the data stream is going/coming from?
Thanks for any help in advanced, and sorry if this has been asked before. I didn't know what I was looking for, so my searching was limited.
Sean _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
