Sip Jitterbuffer is killing asterisk-


I had lots of trouble when activating Sip-Jitterbuffer(adaptive, no force) with FreePBX 2.9rc.
Using the built in Attended-Transfer funktion of our Grandstream GXP2110’s (not the asterisk/freepbx-transfer) simply crashed our Asterisk- (Debian-Lenny) . As this is a live-machine I couldn’t crosscheck this issue with other installations right now. Can anyone tell me if it’s a freepbx, asterisk or even a grandstream issue ? …solved in more recent Asterisk Versions ?

best regards