Follow Me & Single Line VoIP Trunks


I have a voip and PSTN line coming into Asterisk. Both the voip and pstn lines only support a single call active on the line at once.

I use the voip trunk for incoming calls, and have a follow me defined that forwards to my mobile phone if the call is not answered.

This works in terms of connecting the call, but there is no sound at either end of the call.

When I check what happens in the Asterisk console, I can see that the call comes in, is unanswered, then a new call is made outbound to my mobile phone. However, even though I have the channels for both the voip and pstn trunks both set to (1), asterisk still tries to connect the call using the voip channel that the inbound call is coming in on.

I am presuming that while mynetfone (australia) limits the simultaneous calls to (1) for this service, it will still allow the call to be connected, just not pass any sound.

To test this further, I tried the following I make a call outbound on the voip line, then attempt another call out, and the outbound route I have defined states that the voip line should be tried first, and then the pstn. However I get “Call could not be connected”. Suggesting that Asterisk has understood that the voip line is busy, and failed to recognise that the PSTN line is available.

So in the first scenario, it seems to not recognise the channel limit on the voip line, and in the second scenario, it does recognise the limit, but then doesn’t recognise that there are multiple trunks available.

Can any on see what is going on here?

This is freepbx 2.2.1, and asterisk 1.2.14

My dialplan for outbound calls is XXXXXX. and has the voip trunk first in the list, and the pstn zap trunk second.



both asterisk and freepbx. both are somewhat old, and there have been a lot of bugfixes…


unfortunately the call limit on trunks is limitted to outbound calls only. The inbound calls are not counted. To do such would require some custom dialplan on your inbound context.

How the trunk failover works if the first trunk is not availabe will depend on the signalling that is delivered by your voip provider, since each provider handles it differently and some providers send back sip signalling that is interpreted by asterisk as ‘busy’ and not ‘congestion’ - the former will not try the next trunk, the latter will.