I got a real weird one that I want to run by y'all before I give up and buy some support. (And then they give up) I have this odd behavior that only happens for one user (But he is the CEO, of course) and happens often, but is not repeatable. Elastix 2.0.2 fully yum updated. A US T1 trunk into an OpenVox DE115e card. Aastra 6730i sip phone. uLAW and aLAW codecs. Moderatly crowded network... He will be on a call, and the call will drop for him. The other party will still be on the call and unaware. He picks up and dials any number or extension and is dumped back into the call. He can hear the other party, but they can not hear him. No matter what he does (Hangup or stay on) the call will not terminate until the other party ends the call, or I yank the T1 from the card. (What I had to do on a conference call that would not end) Even repeated hangups and a call to voicemail *97 will dump him back in the call. Going into "astersik -r" using "dahdi show channels" will NOT show that line as active. And the old "soft hangup" no longer seems to wok in 2.0.x so I could not do that. Any hints? Should I kill the server before it starts to eat the brains of the mail system? Should I watch "Shaun of the Dead" again?