[asterisk-dev] Found a bug

Fri Sep 14 07:47:48 CDT 2007

If channel clearing problem is solved then I need a patch for this......Can
anybody help me with this.

On 9/14/07, Atis <atis at best.eu.org> wrote:
>
> 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
> > _______________________________________________
> >
> > Sign up now for AstriCon 2007!  September 25-28th.
> http://www.astricon.net/
> >
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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
>
> _______________________________________________
>
> Sign up now for AstriCon 2007!  September 25-28th.
> http://www.astricon.net/
>
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-dev mailing list
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>

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