Help with pstn connection

Dear All

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.

Please can anybody help ?

Awaiting your replies


Dahdi recognizes the modem??

thanks for your reply, how do i check if dahdi recognise the modem pls ?

thanks for your help

Run command:


Yes, I get

PCI:0000:02:0b.0 wcfxo+ e159:0001 wildcard X101P clone

what is next pls ?

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.


Thanks for your help

I didnt configured DAHDI, our numbers are a 10 digit number.

What i need is for example I press a number to get to the outside pstn line through the modem. For incoming I need it to be forwarded on an extension.

Awaiting your replies

thanks and regards

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

In your chan_dadhi put this:

context=incoming callerid=asreceived usecallerid=yes transfer=yes echocancel=yes echocancelwhenbridged=no echotraining=yes rxgain=1.0 txgain=1.0 busydetect=yes busycount=6 signalling=fxs_ks faxdetect=no group=1 dahdichan=1

Then in your dialplan create rules like:

;for incoming calls
exten => s,1,answer()
exten => s,n,background(yourivr)


exten=> s,1,dial(sip/yourextension)

;for outgoings calls
exten=> _9XXXXXXXXXX,1,dial(dahdi/1/${EXTEN:1}[/code]


Ok I managed with system.conf, where should i find chan_dahdi and dial plan conf files?


You will find in /etc/asterisk/

ok all then what is next pls ?


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.

nothing seems to happen is it ok if i pm you my ip for you to have a look pls ?


did you manage to log in pls ?


Nope, I cant see the GUI.

Are you using freepbx or asterisk from scratch?

I only a Hotel page with that IP

oh sorry your are being routed to my test server, try now you should find aterisk gui

i it ok now pls?

Yes I do.

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

still no success :frowning:

my pstn line number is 27007236 …i entered 7236 in DID. and for outgoing what prefix shall i dial to get the outside line ?

please feel free to do a test call and add an extension for testing purpose

thanks for your help