[asterisk-dev] Found a bug

[Rizwan Hisham]

Thats not the problem. I am currently using qualify=yes. But if somebody
does not use it then the call will not go through, thats also fine coz its
firewall problem. But when the call is not successful then the sip channel
should be cleared from active sip dialogues. But it doesnt.

On 9/15/07, Nicholas Blasgen <nicholas at blasgen.com> wrote:
>
> 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/>
> > > > >
> > > > > _______________________________________________
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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 <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
> >
> > _______________________________________________
> >
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>
>
> --
> /Nick
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-- 
Best Regards
Rizwan Hisham
Software Engineer
Axvoice Inc.
www.axvoice.com
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