Ishfaq Malik
2010-Feb-10 14:11 UTC
[asterisk-users] Muted calls occasionally dropping after 30 seconds
Hi 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. 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
Jeff Brower
2010-Feb-10 17:18 UTC
[asterisk-users] Muted calls occasionally dropping after 30 seconds
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