asterisk users - May 2007 - call-limit=2 , call counter not reset to zero after hangup

Hi all,
There is a case in which the call counter is not set to zero for a sip peer
(incoming call). Here is the scenario.

Dialplan:

exten=> 1,1,Dial(SIP/U1)
exten=> 1,2,Gotoif($["${DIALSTATUS}"="ANSWERED"]?:10,1)
exten=> 1,3,Hangup

exten=> 10,1,Voicemail()

If a user just registered with my asterisk and due to some reason after
sometime the user's ATA gets turned off. But asterisk does not know that
becoz there is some time left for the ATA to re-register itself. Now if
anyother user calls U1 he will get a dialstatus of NOANSWER and will be
thrown to voicemail. after voicemail has ended and the calling user has hung
up. the incoming call limit counter for the called user (in this case U1) is
not reset to zero. This only happens in this particular case. otherwise the
call counter for peer and user works fine.

So if anybody knows how to fix this prob, plz share the solution. Can i set
the call counter to zero from the dialplan? or is there anyway to know that
the user is registered or not before dialing that user?

I am using asterisk 1.4.2
-- 
Rizwan Hisham
Software Engineer
AXVOICE Inc.
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20070518/034d9097/attachment.htm