First time user, no SIP, just landline



first time user here (asterisk is installing right now on server behind me (installing from USB so I keep having to hit enter… does this still work?)

thought I should say hi, introduce myself and describe what I’m trying to do… figured you guys might be able to give me a shove in the right direction
(I’ve been googling and reading here for a few weeks in prep)

I don’t need any SIP services right now (or probably ever?)

I just want to plug my land line into the Asterisk server and then be able to use that phone line over the network / local IP phones

might add an extra landline later on if needed

answer machine would be a nice bonus

and caller ID would be great if I can view that remotely on a windows machine / have that machine fetch/display caller info on screen (so when it recognises customers phone number it displays their info - if I can get caller ID remotely I can sort the rest)

I think I should be able to figure it all out with google but will probably post back some questions later on!

if there’s anything big/obvious/handy don;t be afraid to shout at me and tell me what to do




The sky is the limit. You’ll need an FXO card to plug your land line in and an FXS card if you want to use an analog phone (FXS requires power from the computer board, FXO does not). Digium makes cards with replaceable modules. You could have one FXS and one FXO on the same card or multiple of each. You can have Voicemail (answering machine) stored on your computer or email it to yourself. If you have multiple outbound lines, you could forward the call to your cell, or register a sip phone from your cell. The variables for callerid are already built in.

I know RTFM isn’t very nice, but I RTFM twice and went to a training class. Both are very helpful.


hi mkozusink

thanks for the help, I was going to reply earlier but thought I should do a bit more work and follow your advice before posting back!

server is up and running, I’ve followed a few of the basic guides I found online
DXO card installed

I bought a couple of Sangoma phones to use first, figure I can add the cisco ones later on
end point manager is setup, phone is connecting to the server, updating settings etc. through tftp

I’ve left the firewall turned off for now - I don;t need any phones connecting from outside, server will be behind a Pfsence box anyway

I even bought an “asterisk for dummies” book… it’s the first time I’ve ever had to buy a book for something like this… it doesn’t feel quite right not being able to work it out between myself and google :-o

but… i’m still a bit stuck, half the problem is I don’ really know what I should be searching for / reading up on… I spend ages learning something… then find out it’s something I’ll never need
(I have a brain injury, I’m pretty normal but don’ learn new things as fast as I used to)

I’d really like it if… when the land line starts ringing… the asterisk box will make the IP phones ring before it answers the line… is this possible? - if so what’s it called and/or what should I be googling ?

thanks again for the help



I’d really like it if… when the land line starts ringing… the asterisk box will make the IP phones ring before it answers the line… is this possible? - if so what’s it called and/or what should I be googling ?

The simplest, one line, dialplan will do this. As long as you don’t do anything that causes the incoming call to be answered, Dial will not answer it until an outgoing line has answered.


hi David

thanks for the reply

can you tell me how to do that ? I can only find dialplans for outgoing calls, not incoming, not sure if I’m looking in the wrong place or just doing this wrong

EDIT: ok I found it now… ‘Add Inbound DID’ under Advanced tab after I’ve clicked which extension it is right?

EDIT2: edit 2 so I don’t keep bumping the thread… should I upgrade to freepbx 14 now while the system is being used ? - better to get it out of the way now ?


There are no tabs, as there is no GUI. If you are using a GUI, you need to use the GUIs support channel.

I have a feeling that FreePBX answers incoming calls prematurely.

Note that Asterisk does not differentiate between incoming and outgoing calls.

FreePBX peer support is at


ahh :-o

that’ll help a lot

thanks :slight_smile: