Incoming calls via fxo card!

Hey everyone ,

it’s my first time with astrisk, i installed it and created a extensions and it’s work fine & i can access it from outside perfectly , my issue that i attached a openvox card FXO & plug the outside line (( the analog phone line )) okay i created a extension 500 with port one on the card , so the sip phone can call the 500 EX and make an outside calls .until here everything is perfect but how i can receive calls from outside through the FXO card !!! and i want to know can i make the analog phone call the ip phone in the same network i have more than 10 analog phones in my company wiht a 4 ip phone how i can make both of them call together via FOX card plz i need ur help

There is quite a lot of information in configs/extension.config.sample.

For more in depth information see

The doc directory goes into further detail, in specific areas.

The web site has a lot of information, although some of it is unreliable or obsolete.

If you have questions on specific details, you can ask here, but please show what you have tried and at least the verbose logging from it. You will get the best results if you have made simple mistake and it is clear what that is, and can be answered in few words.

If you don’t have time to read the documentation, there is a jobs forum, where you can ask for paid support.

If you have a current genuine Digium card, you may be able to get support from Digium, but I don’t know the limits of that support. Digium do not support their products on this forum.

thanx fro ur replay ,

I’m just need to be sure about one thing if u can replay to me it will be nice ,

the FXO can receives calls and convert it to ip or analog phone without FXS ?!!

The FXO module can receive calls from an analogue PSTN line or an extension (FXS) port on a PABX that implements a standard analogue telephone interface. It can also originate calls on such lines.

In both cases, it makes the call available to the CPU on the host PC. That CPU, and the program it is running, e.g. Astgerisk, is responsible for accessing other hardware, e.g. an ethernet interface, for VoIP phones, or an FXS module, for analogue telephones, and forwarding the speech too and from it.

The FXS interfaces may be part of an analogue telephone adaptor (ATA), which looks like a VoIP phone to the PC.

You probably need to read Chapter 7, in particular: … eConn.html

First off you say you’ve created an extension (500) on your OpenVOX card with an FXO interface. You need to create a trunk to point to the card and it’s FXO interface(s) not an extension. Extensions are for phones (SIP/Analogue/etc). For analogue phones you will need some FXS interfaces provided by either an interface card on the Asterisk server or via some sort of Analogue Telephone Adapter (ATA). To be honest it might cost almost as much to buy ATA’s as it would to simply replace the old analogue phones with IP handsets.

Thanx to both of u ,

but let me make my idea more clear , i have my PBX with one FXO port this port connect to a Panasonic system which have a many of analog phones and two PSTN lines okay , as i said i created a EX 500 as Zap when i calling it from sip phone it give me the line that on a Panasonic system & i can call any analog phone or make a outside call .

what i want to do now to make a call from analog phone to sip phone both of them r in the internal network connected together through the FXO port .

Can I DO That ? & if Yes How ?

Yes you can do that.

Its work for me for following scenario

jitsi---->asterisk----------->analog PBX----------> Telephone
and vice versa also

Instead of jisti you can use your voip phone. I use jitsi in my PC.

for that you have to configure your extension.conf file. Before that also have to configure chan_dahdi.conf file to set your TDM card.

Note that “zap” is obsolete. Otherwise this should be a very normal configuration for which peer support is inappropriate until you hit a specific problem.