[asterisk-dev] Found a bug

Fri Sep 14 18:37:02 CDT 2007

Rizwan,

Can you please try it with qualify=yes ?  Qualify will insure that the
firewall keeps it's ports open.  Somehow I don't think this is going to be a
bug with Asterisk.

Atis,

If you're using RT you can turn on RTCacheFriends and just issue a "asterisk
-rx sip reload" or it's something like that.  The other option is to use the
Asterisk Manager Interface (AMI) to issue a CLI "COMMAND" to reload SIP (or
whatever channel driver).  I think there is also another way to tell
RealTime to clear it's cache but I can't think of it off the top of my
head.  Oh, and by turning on RTCacheFriends you'll have access to
"qualify=yes".

On 9/14/07, Rizwan Hisham <rizwanhasham at gmail.com> wrote:
>
> 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 <http://www.best.eu.org/>
> > > >
> > > > _______________________________________________
> > > >
> > > > Sign up now for AstriCon 2007!  September 25-28th.
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> > >
> > >
> > >
> > > --
> > > Best Regards
> > > Rizwan Hisham
> > > Software Engineer
> > > Axvoice Inc.
> > > www.axvoice.com
> > > _______________________________________________
> > >
> > > Sign up now for AstriCon 2007!  September 25-28th.
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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 <http://www.best.eu.org/>
> >
> > _______________________________________________
> >
> > Sign up now for AstriCon 2007!  September 25-28th.
> > http://www.astricon.net/
> >
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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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>
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-- 
/Nick
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