Ishfaq- > I'm having a very odd phenomenon happening on our production server > (1.4.17 and using realtime). Sometimes a call will disconnect 30 seconds > after the SIP phone hits the mute button but it doesn't happen all the > time. I've done a sip debug while watching this happen and that doesn't > show anything other than a BYE message being sent out of the blue.
Are you using a codec (such as G729) on the outgoing leg of that line? If so you might check for VAD/DTX enabled and see if that makes any difference. -Jeff > The rtptimeout and rtpholdtimeout are both set to 0 on a global level > and for the sip extension the sip table row has NULL in both columns. > > I've tried playing with those 2 values, both on a global and sip > extension level but regardless to what they are set to, if the call gets > disconnected it is always 30 seconds after the mute button is pressed. > But like I said before, this does not happen every time the mute button > is pressed. > > I managed to recreate the phenomenon one one of our test servers so I > could be certain that there was nothing else going on at the time. > > The call path when recreating this on our test platform was My Mobile -> > number/SIP provider -> out asterisk server -> SIP extension > > Has anyone else ever experienced anything like this? It's really got me > rather frustrated! > > Thanks in advance > > Ish > -- > Ishfaq Malik > Software Developer > PackNet Ltd > > Office: 0161 660 3062 -- _____________________________________________________________________ -- 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