[asterisk-users] Muted calls occasionally dropping after 30 seconds

[Ishfaq Malik]

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