I just installed asterisk and it looks like a great piece of software. I managed to configure most of the things and extensions with voip are working correctly (through sip softphones).
However I am having problems on how to configure FXO, trunking, dialing plan for the external pstn line.
I am testing with an INTEL MD3200 modem before purchasing a genuine fxo.
The next step is configure your /etc/dahdi/system.conf and /etc/asterisk/chan_dahdi.conf. If you have configured yet, please show us with the rules in your dialplan. if not run dahdi_genconf and then check this.
Ok, you first need to configure dahdi beacuse asterisk use the configuration for dial, in your /etc/dahdi/system.conf put this:
fxsks=1 ;here you define a FXO port
echocanceller=mg2,1 ; here you define a echo cancelelr for that port
loadzone = us ;here you define your zone
defaultzone = us
You need to relaod the diaplan from asterisk CLI running the command: dialplan relod, then just use one of your phones to generate a outgoing call and with your cell phone or othr phone call to the line number that you are connecting.
You will hear your IVR if you create one or rng the extension that you defined for incoming call. And the outgoing must be answered. For all put the output CLI here.
Now forget the part on exensions.conf, I thought you are using asterisk from scratch. For freepbx is tiny different.
In the section of Inbound routes in your Test configuration add the last 4 numbers of your line number. Per exampe if your line number is 5556789900 put in the DID number case only 9900 and save it, already you have a destination.
In the outbound routes you have two rules NXXNXXXXXX and NXXXXXX this will only permit dial numbers for [2-9]XX[2-9]XXXXXX and [2-9]XXXXXX check if this accord you want.
And try again to dial out and dial in. In both cases put the output cli here for more debugging