[asterisk-dev] Found a bug

[Atis]

On 9/14/07, Rizwan Hisham <rizwanhasham at gmail.com> wrote:
> Well my main problem is that sip channels get stuck, "sip show channels"
> shows stucck channels in initial invite state while "core show channels"
> show nothing related to those channels. Do you see the same? i guess if the
> channel is cleared then the call limit will also be updated automatically.

Nop, i don't see any channels left. This probably is fixed, i'm using 1.4.10.

> Turning the linksys off may not be the best idea for NAT simulation. Try
> unplugging you lan cable from it unplug your telephone cable out from your
> main modem just after registering and then try it.

And i just saw an answer - why qualify isn't working for me - because
i'm using RT. I just ton't get why it's so - a registered SIP phone
can be cached in asterisk until registration times out.. and if
asterisk detects connection problem, it can update registration
timeout to past value.

Regards,
Atis

> In my case qualify seems to be working fine.
>
>
> On 9/13/07, Atis <atis at best.eu.org> wrote:
> >
> > 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
> >
> > --
> > Atis Lezdins,
> > IT Responsible of BEST Riga,
> > atis at BEST.eu.org
> > ICQ: 142239285
> > Skype: atis.lezdins
> > Cell Phone: +371 28806004 [Tele2, Latvia]
> > Work phone: +1 800 7502835 [Toll free, USA]
> > ?BEST? -> www.BEST.eu.org
> >
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>
>
>
> --
> Best Regards
> Rizwan Hisham
> Software Engineer
> Axvoice Inc.
> www.axvoice.com
> _______________________________________________
>
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-- 
Atis Lezdins,
IT Responsible of BEST Riga,
atis at BEST.eu.org
ICQ: 142239285
Skype: atis.lezdins
Cell Phone: +371 28806004 [Tele2, Latvia]
Work phone: +1 800 7502835 [Toll free, USA]
?BEST? -> www.BEST.eu.org