Hello, worst aspect is that - if SIP clients do not have such a timeout, and in that case if killing an asterisk and to start it up again - so it is nothing to do with this asterisk timeout.
Regards, On 23 February 2010 08:44, Olle E. Johansson <o...@edvina.net> wrote: > > 23 feb 2010 kl. 01.47 skrev Kirill 'Big K' Katsnelson: > > > On 100222 1313, JT wrote: > >> When a SIP device dials another SIP device...Asterisk connects the calls > and > >> displays the channel information. > >> If one of those SIP devices hangs up, Asterisk receives the hangup > notice > >> and disconnects the call/channel. > >> However - what does Asterisk do when the network cable is unplugged from > one > >> of the SIP devices...?! > > > > Jared already mentioned SIP session timers, which are supported starting > with 1.6. Here's my experience. While I am running 1.6, the software stack > that is used for agent softphone (PJSIP) does not support the session > timers. If the softphone crashes in a call, the call would get stuck exactly > as you describe. > > > > I am working around this problem by setting rtp timeouts in sip.conf: > > > > [general] > > rtptimeout=10 > > rtpholdtimeout=300 > > > > This means that if RTP flow stops while the agent is in the call, the > call will be disconnected in 10 seconds. If the call was put on hold by the > agent, it will be disconnected in 300 seconds. Your timeouts may vary. > > > > The caveat here is that it is perfectly normal NOT to transmit any RTP > data in case of long silence. > Not in Asterisk - we do not really support silence suppression. The > recommendation is to turn it off on the phones. > > > This is why the SIP timers were introduced in the first place: there is > no correct way to detect when the client is going away, as no activity is a > good session state. > > > > I am able to get away with the small timeout because I set the PJSIP > client to always transmit RTP, by turning off voice activity detection > feature (VAD). If you want to support that feature, set rtptimeout as high > as for how long you allow absolute silence on the line without disconnecting > it. > > Just to complete this discussion - we also have the absolute timeout that > is a lifesaver in many cases. If you set this to a time that's larger than > the normal calls, Asterisk will hang up the call. I very often set it to two > hours, just to make sure that if anything strange happens, all calls will be > cancelled out at some point. > > /O > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- We never did too much talking anyway So don't think twice, it's all right ---------------------------------------------------------- There are more things in heaven and earth, Horatio, Than are dreamt of in your philosophy.
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