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

[Jeff Brower]

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