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.
<br><br>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.<br><br>In my case qualify seems to be working fine.
<br><br><div><span class="gmail_quote">On 9/13/07, <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 9/13/07, 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 a
<br>> NAT problem with a user and a call comes for that user, asterisk throws an<br>> initial invite towards that user but gets no response from him even after 5<br>> retries. Caller hears nothing.<br>><br>> During this process the call limit is updated and increased for the callee
<br>> and a channel is also created. But after the caller hangsup the call, call<br>> limit is not updated back to zero for callee and 'sip show channels' shows<br>> the callee's channel stuck in an initial invite state. 'core show 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, so<br>> if there is NAT problem for any user then after trying to reach him for 2
<br>> times, his call-limit is reached and rest of incoming calls for him go 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 times
<br>> as i want. Maybe its fixed in new releases of asterisk but havent tried 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 any<br>> user is unreachable then asterisk saves its status UNREACHABLE and 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 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.
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<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>