Unable to recieve incoming calls on my analog lines


I have been working with one of the sales engineers to get my service up and running. This is a strict home service in which I work from home a lot and using it to learn. My issue is as follows.

I have the test SIP trunks up, and i can send and receive calls from my VoIP phone.

I have a Digicom TDP410P 1FSX/3FXO card. I only use the FSX port, and my analog line connects to it.

I can make outbound calls on the analog line. I however cannot receive any inbound calls. I have even picked up the VoIP phone to make a call and pick up my analog phone to see if I could here anything. It was a shot in the dark.

My service over all is, all my analog lines in my private home collapse into a single link into the FXS port on the analog card. I have my VoIP phone sharing the same phone number, I only want one phone number for my entire home.

I have a ring group set up that has my 2 extensions, one a SIP extension and the other is a a DAHDI extenstion.

I am hoping someone can walk me through, or point me in the right direction on how to get all the phones to receive incoming calls.

Thank you


Generally PSTN phone lines will need to connect to an FXO port to work, please elucidate

So to be clear here. I know about that, and that is not how i want it set up.

I have the FXS port in use to all my analog home phones. That line can make outbound calls over the SIP trunk.

The VoIP phone receives inbound and can make outbound calls.

I have a ring group (202) set up catching the extensions 200 (VoIP) and 201 (analog phones).

I have been wondering if I need to create a dahdi trunk for the inbound calls but that module seems to be broke.

Honestly this is my first attempts to do voice. I am a Senior Network designer for company and while I know the data transport for Voice inside and out and am familiar with all the terms. All the forums, videos, training stuff has not shown me how to fully do analog yet.

Hence my question, how to get inbound calls to my entire ring group. IE my analog phones.

Sorry I’m still not understanding, are you saying you have all your phones connected in parallel to themselves and also to the incoming PSTN " Phone Line" and also to an FXS port?


Think of your normal house, i just work a lot from. All the analog phones in your house come to one entry point.
Just think of this as one phone period. One analog line that connects to the FXS port.

I can make outbound calls no problem from that analog FXS connection.

I want to receive inbound calls on that analog. I want the VoIP phone and my analog phone to use the single DID and to make/receive calls on it.

I believe you confirmed my suspicion.

To extend your analogy, and because you are a network engineer a normal house is like an old fashioned 10base hub where everything is bridged together, however a PBX is more like a router that can functionally replace that hub and add facility, as such you attach the incoming line(s) from your telco (a current source) to FXO port(s) attached to the PBX, the phones themselves (current sinks) attache preferably individually to FXS ports, you can “bridge” several phones together to act as on device if you keep the resultant Ringer Equivalency below that of the FXS port.

Then to do what you want, you just setup trunks, inbound and outbound routes etc. just as the wiki shows

Have all of that set up. As I stated, I can do everything but one thing. That is receive incoming calls to the analog phone.

Your analogy is correct.

I also suspect that since I cannot create a dahdi trunk I think that is my root issue. From what I am reading since yesterday is that the module is broken and no one can create them at the moment.

Not quite all, you don’t have the FXO trunk yet, try again when you fix that.

I do not want the FXO port. I want to route the analog out the SIP trunk. I should have been a lot clearer but having walking pneumonia at the moment is not helping.

I assume you mean the phones attached to the FXS port by “the analog” if so That is handled by your outbound routes, your ring groups and the assignment of the FXS port to an internal extention.

So to restate my issue. I can do everything but receive inbound calls on the analog line.

It will be impossible to help you if you don’t exactly define what is your “analog line” , if it is a connection on a pair of wires to a PSTN provider which reacts to another person dialing that “number” from somewhere else and which terminates in your house and makes pieces of hardware ring , you will NEED an FXO trunk, directly connected to a hardware port available to your PBX, if not you need to explain what you exactly mean, if you just mean the parallel connections of you current “phones” connected to an FXS port on your PBX, then just direct the inbound SIP call to ring the extension you have defined as that FXS.

The confusion in this thread stems from the fact that you are calling your analog phone a “line”. You have probably misconfigured your DAHDI Extension, so to test that theory we need call traces for two cases:

  1. call from your DAHDI Extension to your softphone extension
  2. call from your softphone ext to the DAHDI ext.

How to get a call trace:

i now have all my services up and going. I am sorry i could not convey voice world terminology. I am a data engineer who rarely does this. I ended up having quite a few corruptions on the FREEPBX install. No idea why or how it happened. Since I signed up for the test service first, and tried to override with the new settings, I must have done a lot wrong.

It took 2 days to rebuild, no idea why the install media froze constantly. I had 1 install work out of 57 attempts in 2.5 days now. However, i learned enough to rebuild it, configure my new trunks. Set up DAHDI and my DAHDI extension, My VoIP phone, and its Extension, and my ring group. I can now make calls from any analog phone in my home or my VoIP phone and I can receive calls on all my analog phones and my VoIP phone.

Now its time to play with voicemail, fax, and my video part of my phone. Thanks for the insight and terminology. I can see i have quite a bit to learn about the voice side.

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Just an FYI i figured out why my incoming calls did not go to my FXS port. When I originally configured the box, I had an issue with the TDP410P and the FXS port did not come up. It was on port 1, and to troubleshoot we moved the daughter board. Once we figured out i had an issue with the molex cable, we never moved it back. The FREEPBX config kept referencing channel 1. I thought it was odd that i could see in DAHDI that the FXS port was identified as 2, but yet when i configured my DAHDI extension It kept trying to use channel 1.

Maybe this could be use for someone else that experiences what I went through. Thanks again for all the insight.

no, that is totally normal, if your fxs extension dosn’t point to the correct hardware, it just won’t work. the dahdi helper is less than fully reliable, and if course didn’t know when you subverted it , just delete your previous work and start over next time.

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