On 9/13/07, Rizwan Hisham <[EMAIL PROTECTED]> 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, [EMAIL PROTECTED] ICQ: 142239285 Skype: atis.lezdins Cell Phone: +371 28806004 [Tele2, Latvia] Work phone: +1 800 7502835 [Toll free, USA] ?BEST? -> www.BEST.eu.org _______________________________________________ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
