i am using X100P on RHEL4, all incoming calls doing well, during any outbound call from sip to pstn, it hangup right away when the remote side pick up the phone. i've been trying to trace out this problem for 2days. for the log snapshot below, DEBUG[2401]: Exception on 15, channel 1 DEBUG[2401]: Got event On hook(1) on channel 1 (index 0) the On hook event always happens when the remote user pick up the phone. that's mean when i doing outbound call and the remote user did not pick up the phone(that is the phone keep ringing) it won't drop off. to my understanding on hook should mean the remote side pick up the phone. this On hook event should be handled correctly by the hardware right? but then asterisk drop the connection right away. i have no problem running with the same hardware on centos 3(kernel 2.4) do you think it's related to any asterisk problem on kernel 2.6.9(RHEL4)? -vince DEBUG[2401]: Exception on 15, channel 1 DEBUG[2401]: Got event Hook Transition Complete(12) on channel 1 (index 0) DEBUG[2401]: Exception on 15, channel 1 DEBUG[2401]: Got event Dial Complete(9) on channel 1 (index 0) DEBUG[2401]: No echocancellation requested DEBUG[2401]: Dropping duplicate answer! VERBOSE[2401]: -- Zap/1-1 answered SIP/168-9dbb DEBUG[2401]: Ooh, format changed from unknown to ulaw DEBUG[2401]: Stopping retransmission on '541148A8-3E85-4997-A415-785BE63E8186@10.10.10.103' of Response 10010: Found DEBUG[2401]: Exception on 15, channel 1 DEBUG[2401]: Got event On hook(1) on channel 1 (index 0) DEBUG[2401]: Didn't get a frame from channel: Zap/1-1 DEBUG[2401]: Bridge stops bridging channels SIP/168-9dbb and Zap/1-1 DEBUG[2401]: Hangup: channel: 1 index = 0, normal 15, call wait = -1, thirdcall = -1 DEBUG[2401]: Set option TDD MODE, value: OFF(0) on Zap/1-1 DEBUG[2401]: Updated conferencing on 1, with 0 conference users