Hi all,<br>There is a case in which the call counter is not set to zero for a sip peer (incoming call). Here is the scenario.<br><br>Dialplan:<br><br>exten=> 1,1,Dial(SIP/U1)<br>exten=> 1,2,Gotoif($["${DIALSTATUS}"="ANSWERED"]?:10,1)
<br>exten=> 1,3,Hangup<br><br>exten=> 10,1,Voicemail()<br><br>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.
<br><br>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?<br><br>
I am using asterisk 1.4.2<br>-- <br>Rizwan Hisham<br>Software Engineer<br>AXVOICE Inc.