All of our outgoing phone calls go via a local asterisk server connected to Internode’s voip service NodePhone. One problem we’ve had is that muting our phone line for long periods of time, which I like to do during conference calls, would lead to the call being disconnected. I should complain more often on twitter as someone suggested I look into RTP keep alive.
By default asterisk does not terminate calls if there is no RTP activity, but neither does it send keep alive packets to the voip service provider if no packets are received when someone mutes their line. Adding the following to the general section of
sip.conf fixed the call dropout problem: