[asterisk-dev] Found a bug

[Atis]

On 9/13/07, Rizwan Hisham <rizwanhasham at gmail.com> wrote:
> 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.

Hi,

I'm not sure is this related or not, but i have few Linksys PAP2
devices behind NAT, that regularly get disconnected from asterisk.
Symptoms are the same - after few calls (not necessarily 2, however my
call-limit is also 2) i hear silence after Dial().

I just tried testing, but doesn't seem that qualify=yes helps in any
way. Maybe i'm not simulating NAT problem correctly? Or is it bug in
qualify setting? I'm just powering off linksys, and i'm hearing
silence. Shouldn't qualify=yes almost immediately mark device as
UNREACHABLE?

Regards,
Atis

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Atis Lezdins,
IT Responsible of BEST Riga,
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