On 6/7/07, Douglas Garstang <[EMAIL PROTECTED]> wrote:
That's all fine and good until it becomes the receiving phone, and the other phone (as an originator) also has canreinvite set to yes. Then, your back to both Asterisk servers being completely taken out of the loop again!
While I haven't taken the time to actually try this, I might suggest that you could set up separate user and peer sections in sip.conf, so that you can handle inbound calls differently that outbound calls. -Jared _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
