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.
<br><br><div><span class="gmail_quote">On 9/15/07, <b class="gmail_sendername">Nicholas Blasgen</b> <<a href="mailto:nicholas@blasgen.com">nicholas@blasgen.com</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;">
<div>Rizwan,</div>
<div> </div>
<div>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.</div>
<div> </div>
<div>Atis,</div>
<div> </div>
<div>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".
<br><br> </div>
<div><div><span class="e" id="q_115066f674f298d2_1"><span class="gmail_quote">On 9/14/07, <b class="gmail_sendername">Rizwan Hisham</b> <<a href="mailto:rizwanhasham@gmail.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
rizwanhasham@gmail.com</a>> wrote:</span>
</span></div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0px 0px 0px 0.8ex; padding-left: 1ex;"><div><span class="e" id="q_115066f674f298d2_3">If channel clearing problem is solved then I need a patch for this......Can anybody help me with this.
<div><span><br><br>
<div><span class="gmail_quote">On 9/14/07, <b class="gmail_sendername">Atis</b> <<a href="mailto:atis@best.eu.org" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)"> 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" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
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" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
atis@best.eu.org</a>> wrote:<br>> ><br>> > On 9/13/07, Rizwan Hisham <<a href="mailto:rizwanhasham@gmail.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">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" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">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/" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
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/" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
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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/" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
www.axvoice.com</a><br>> _______________________________________________ <br>><br>> Sign up now for AstriCon 2007! September 25-28th. <a href="http://www.astricon.net/" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
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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" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
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/" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
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</a><br></blockquote></div><br><br clear="all"><br></span></div>--
<div><span><br>Best Regards<br>Rizwan Hisham<br>Software Engineer <br>Axvoice Inc.<br><a href="http://www.axvoice.com/" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
www.axvoice.com</a> </span></div><br></span></div>_______________________________________________<span class="q"><br><br>Sign up now for AstriCon 2007! September 25-28th. <a href="http://www.astricon.net/" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
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</a><br></span></blockquote></div><span class="sg"><br><br clear="all"><br>-- <br>/Nick
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</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>