[asterisk-dev] Found a bug

[Rizwan Hisham]

I have sip users with the following configuration:
[abc]
username=abc
type=friend
secret=123
qualify=no
nat=yes
insecure=port,invite
call-limit=2
host=dynamic
dtmfmode=rfc2833
context=uscan
canreinvite=yes

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.

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.

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.

Can somebody help me fix this problem?

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.







-- 
Best Regards
Rizwan Hisham
Software Engineer
Axvoice Inc.
www.axvoice.com
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