If channel clearing problem is solved then I need a patch for this......Can anybody help me with this.<br><br><div><span class="gmail_quote">On 9/14/07, <b class="gmail_sendername">Atis</b> <<a href="mailto:atis@best.eu.org">
atis@best.eu.org</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">On 9/14/07, Rizwan Hisham <<a href="mailto:rizwanhasham@gmail.com">
rizwanhasham@gmail.com</a>> wrote:<br>> Well my main problem is that sip channels get stuck, "sip show channels"<br>> shows stucck channels in initial invite state while "core show channels"<br>
> show nothing related to those channels. Do you see the same? i guess if the<br>> channel is cleared then the call limit will also be updated automatically.<br><br>Nop, i don't see any channels left. This probably is fixed, i'm using
1.4.10.<br><br>> Turning the linksys off may not be the best idea for NAT simulation. Try<br>> unplugging you lan cable from it unplug your telephone cable out from your<br>> main modem just after registering and then try it.
<br><br>And i just saw an answer - why qualify isn't working for me - because<br>i'm using RT. I just ton't get why it's so - a registered SIP phone<br>can be cached in asterisk until registration times out.. and if
<br>asterisk detects connection problem, it can update registration<br>timeout to past value.<br><br>Regards,<br>Atis<br><br>> In my case qualify seems to be working fine.<br>><br>><br>> On 9/13/07, Atis <<a href="mailto:atis@best.eu.org">
atis@best.eu.org</a>> wrote:<br>> ><br>> > On 9/13/07, Rizwan Hisham <<a href="mailto:rizwanhasham@gmail.com">rizwanhasham@gmail.com</a>> wrote:<br>> > > I have sip users with the following configuration:
<br>> > > [abc]<br>> > > username=abc<br>> > > type=friend<br>> > > secret=123<br>> > > qualify=no<br>> > > nat=yes<br>> > > insecure=port,invite<br>> > > call-limit=2
<br>> > > host=dynamic<br>> > > dtmfmode=rfc2833<br>> > > context=uscan<br>> > > canreinvite=yes<br>> > ><br>> > > User registers with asterisk without any problem, but whenever there is
<br>> a<br>> > > NAT problem with a user and a call comes for that user, asterisk throws<br>> an<br>> > > initial invite towards that user but gets no response from him even<br>> after 5<br>> > > retries. Caller hears nothing.
<br>> > ><br>> > > During this process the call limit is updated and increased for the<br>> callee<br>> > > and a channel is also created. But after the caller hangsup the call,<br>> call<br>
> > > limit is not updated back to zero for callee and 'sip show channels'<br>> shows<br>> > > the callee's channel stuck in an initial invite state. 'core show<br>> channels'<br>
> > > does not show any active calls or channels.<br>> > ><br>> > > This is a serious problem for me as i have call-limit=2 for every user,<br>> so<br>> > > if there is NAT problem for any user then after trying to reach him for
<br>> 2<br>> > > times, his call-limit is reached and rest of incoming calls for him go<br>> to<br>> > > voicemail.And evrytime some tries to call him leaves a stuck channel in<br>> > > initial invite state. Im sure this is a bug as i can repeat it as many
<br>> times<br>> > > as i want. Maybe its fixed in new releases of asterisk but havent tried<br>> any<br>> > > new release. I am using asterisk 1.4.2.<br>> > ><br>> > > Can somebody help me fix this problem?
<br>> > ><br>> > > There is a temporary cure for this problem. if i set qualify=yes, then<br>> > > asterisk keeps checking whether all the users are reachable or not. If<br>> any<br>> > > user is unreachable then asterisk saves its status UNREACHABLE and
<br>> whenever<br>> > > a calls come in for that user asterisk does not bother to send any sip<br>> > > packets to that user. Ultimately no channel is created for that call so<br>> no<br>> > > need to increment or decrement cal l limit.
<br>> ><br>> > Hi,<br>> ><br>> > I'm not sure is this related or not, but i have few Linksys PAP2<br>> > devices behind NAT, that regularly get disconnected from asterisk.<br>> > Symptoms are the same - after few calls (not necessarily 2, however my
<br>> > call-limit is also 2) i hear silence after Dial().<br>> ><br>> > I just tried testing, but doesn't seem that qualify=yes helps in any<br>> > way. Maybe i'm not simulating NAT problem correctly? Or is it bug in
<br>> > qualify setting? I'm just powering off linksys, and i'm hearing<br>> > silence. Shouldn't qualify=yes almost immediately mark device as<br>> > UNREACHABLE?<br>> ><br>> > Regards,
<br>> > Atis<br>> ><br>> > --<br>> > Atis Lezdins,<br>> > IT Responsible of BEST Riga,<br>> > <a href="mailto:atis@BEST.eu.org">atis@BEST.eu.org</a><br>> > ICQ: 142239285<br>> > Skype:
atis.lezdins<br>> > Cell Phone: +371 28806004 [Tele2, Latvia]<br>> > Work phone: +1 800 7502835 [Toll free, USA]<br>> > ?BEST? -> <a href="http://www.BEST.eu.org">www.BEST.eu.org</a><br>> ><br>> > _______________________________________________
<br>> ><br>> > Sign up now for AstriCon 2007! September 25-28th.<br>> <a href="http://www.astricon.net/">http://www.astricon.net/</a><br>> ><br>> > --Bandwidth and Colocation Provided by <a href="http://www.api-digital.com--">
http://www.api-digital.com--</a><br>> ><br>> > asterisk-dev mailing list<br>> > To UNSUBSCRIBE or update options visit:<br>> > <a href="http://lists.digium.com/mailman/listinfo/asterisk-dev">http://lists.digium.com/mailman/listinfo/asterisk-dev
</a><br>> ><br>><br>><br>><br>> --<br>> Best Regards<br>> Rizwan Hisham<br>> Software Engineer<br>> Axvoice Inc.<br>> <a href="http://www.axvoice.com">www.axvoice.com</a><br>> _______________________________________________
<br>><br>> Sign up now for AstriCon 2007! September 25-28th. <a href="http://www.astricon.net/">http://www.astricon.net/</a><br>><br>> --Bandwidth and Colocation Provided by <a href="http://www.api-digital.com--">
http://www.api-digital.com--</a><br>><br>> asterisk-dev mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> <a href="http://lists.digium.com/mailman/listinfo/asterisk-dev">http://lists.digium.com/mailman/listinfo/asterisk-dev
</a><br>><br><br><br>--<br>Atis Lezdins,<br>IT Responsible of BEST Riga,<br><a href="mailto:atis@BEST.eu.org">atis@BEST.eu.org</a><br>ICQ: 142239285<br>Skype: atis.lezdins<br>Cell Phone: +371 28806004 [Tele2, Latvia]<br>
Work phone: +1 800 7502835 [Toll free, USA]<br>?BEST? -> <a href="http://www.BEST.eu.org">www.BEST.eu.org</a><br><br>_______________________________________________<br><br>Sign up now for AstriCon 2007! September 25-28th.
<a href="http://www.astricon.net/">http://www.astricon.net/</a><br><br>--Bandwidth and Colocation Provided by <a href="http://www.api-digital.com--">http://www.api-digital.com--</a><br><br>asterisk-dev mailing list<br>To UNSUBSCRIBE or update options visit:
<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-dev">http://lists.digium.com/mailman/listinfo/asterisk-dev</a><br></blockquote></div><br><br clear="all"><br>-- <br>Best Regards<br>Rizwan Hisham<br>Software Engineer
<br>Axvoice Inc.<br><a href="http://www.axvoice.com">www.axvoice.com</a>