[Asterisk-Users] Zap event On hook(1) handling problem

Vincent vincent at yoric.com
Mon Apr 25 00:55:55 MST 2005


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 at 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



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