I have sip users with the following configuration:<br>[abc]<br>username=abc<br>type=friend<br>secret=123<br>qualify=no<br>nat=yes<br>insecure=port,invite<br>call-limit=2<br>host=dynamic<br>dtmfmode=rfc2833<br>context=uscan
<br>canreinvite=yes<br><br>User registers with asterisk without any problem, but whenever there is a NAT problem with a user and a call comes for that user, asterisk throws an initial invite towards that user but gets no response from him even after 5 retries. Caller hears nothing.
<br><br>During this process the call limit is updated and increased for the callee and a channel is also created. But after the caller hangsup the call, call limit is not updated back to zero for callee and 'sip show channels' shows the callee's channel stuck in an initial invite state. 'core show channels' does not show any active calls or channels.
<br><br>This is a serious problem for me as i have call-limit=2 for every user, so if there is NAT problem for any user then after trying to reach him for 2 times, his call-limit is reached and rest of incoming calls for him go to
voicemail.And evrytime some tries to call him leaves a stuck channel in initial invite state. Im sure this is a bug as i can repeat it as many times as i want. Maybe its fixed in new releases of asterisk but havent tried any new release. I am using asterisk
1.4.2.<br><br>Can somebody help me fix this problem?<br><br>There is a temporary cure for this problem. if i set qualify=yes, then asterisk keeps checking whether all the users are reachable or not. If any user is unreachable then asterisk saves its status UNREACHABLE and whenever a calls come in for that user asterisk does not bother to send any sip packets to that user. Ultimately no channel is created for that call so no need to increment or decrement cal l limit.
<br><br><br><br><br><br><br clear="all"><br>-- <br>Best Regards<br>Rizwan Hisham<br>Software Engineer<br>Axvoice Inc.<br><a href="http://www.axvoice.com">www.axvoice.com</a>