# Newbie, Can't get my TDM400P to respond? **URL:** \ **Category:** Asterisk Support\ **Created:** [November 9, 2005, 4:35am UTC](https://community.asterisk.org/t/newbie-cant-get-my-tdm400p-to-respond/2348 "2005-11-09T04:35:17Z")\ **Posts on this page:** 7\ **Page:** 1
**Author:** ![yanicg](https://avatars.discourse-cdn.com/v4/letter/y/c77e96/32.png) [@yanicg](https://community.asterisk.org/u/yanicg)\ **Post date:** [November 9, 2005, 4:35am UTC](https://community.asterisk.org/t/newbie-cant-get-my-tdm400p-to-respond/2348/1 "2005-11-09T04:35:17Z")
Hello everyone. I’m fairly new to this asterisk world. I was able to get a test Asterisk@home system but that was not good enough. I wanted to learn the good stuff. I just bought the O’Reilly Asterisk: The Future of Telephony book. Right off the bat, I am running into difficulty. I have a TDm400P with one FXO and one FXS. I actually have more but I took them off. I wanted to go through the entire book as they demonstrate. In the first Few Chapters, you setup both FXO and FXS. I should be able to call that phone number and the system should pick up and start the Echo program…phone just rings and rings. Also, when I plug an analog phone into the system, get no dial tone and no response when I dial 611. Here are my settings. Can someone help guide me where I went wrong here? I would really apreciate it! Thanks in advance. BTW, the FXS module is in port 1 and FXO in port 2 and I do hear the beeps from the phone buttons dialing. Just nothing happens. zaptel.conf fxoks=1 fxsks=2 loadzone=us defaultzone=us zapata.conf [channels] usecallerid=yes hidecallerid=no callwaiting=no threewaycalling=yes transfer=yes echocancel=yes echotraining=yes immediate=no ;define channels context=internal signalling=fxo\_ks channel =\> 1 context=incoming signalling=fxs\_ks channel =\> 2 extensions.conf [incoming] ;incoming call from FXO port are directed to this context from zapata.conf exten =\> s,1,Answer() exten =\> s,2,Echo() [internal] exten =\> 611,1,Answer() exten =\> 611,2,Echo() # Zaptel Configuration Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2 channels configured. ---
**Author:** ![kisong](https://avatars.discourse-cdn.com/v4/letter/k/90db22/32.png) [@kisong](https://community.asterisk.org/u/kisong)\ **Post date:** [November 9, 2005, 8:51am UTC](https://community.asterisk.org/t/newbie-cant-get-my-tdm400p-to-respond/2348/2 "2005-11-09T08:51:02Z")
what type of respond do you want? ---
**Author:** ![yanicg](https://avatars.discourse-cdn.com/v4/letter/y/c77e96/32.png) [@yanicg](https://community.asterisk.org/u/yanicg)\ **Post date:** [November 9, 2005, 5:46pm UTC](https://community.asterisk.org/t/newbie-cant-get-my-tdm400p-to-respond/2348/3 "2005-11-09T17:46:46Z")
Well, from what I understand, Asterisk should pick up the call and perform an Echo test. On the phone end, shouldn’t I get a dial tone, be able to input 611 and the Echo test starts? ---
**Author:** ![yanicg](https://avatars.discourse-cdn.com/v4/letter/y/c77e96/32.png) [@yanicg](https://community.asterisk.org/u/yanicg)\ **Post date:** [November 10, 2005, 4:41am UTC](https://community.asterisk.org/t/newbie-cant-get-my-tdm400p-to-respond/2348/4 "2005-11-10T04:41:17Z")
Should I be providing more details here? I don’t know. This seems like something not too complicated. I would really like to get this to work. ---
**Author:** ![angler](https://avatars.discourse-cdn.com/v4/letter/a/96bed5/32.png) [@angler](https://community.asterisk.org/u/angler)\ **Post date:** [November 10, 2005, 11:01pm UTC](https://community.asterisk.org/t/newbie-cant-get-my-tdm400p-to-respond/2348/5 "2005-11-10T23:01:03Z")
This all looks just fine. If you are still having issues please call Digium Support. ---
**Author:** ![ellives](https://avatars.discourse-cdn.com/v4/letter/e/919ad9/32.png) [@ellives](https://community.asterisk.org/u/ellives)\ **Post date:** [November 11, 2005, 1:36am UTC](https://community.asterisk.org/t/newbie-cant-get-my-tdm400p-to-respond/2348/6 "2005-11-11T01:36:45Z")
> [@angler](#): > > This all looks just fine. If you are still having issues please call Digium Support. Did you use AMP to “Add a trunk” and choose the “incoming calls” menu choice to handle what happens with the trunk? Seems like a good way to understand what happens with the .conf files. ---
**Author:** ![yanicg](https://avatars.discourse-cdn.com/v4/letter/y/c77e96/32.png) [@yanicg](https://community.asterisk.org/u/yanicg)\ **Post date:** [November 11, 2005, 11:20pm UTC](https://community.asterisk.org/t/newbie-cant-get-my-tdm400p-to-respond/2348/7 "2005-11-11T23:20:24Z")
Acutally. I don’t have anything set under the Trunk section. lol, the book never mentioned to do that but that could be the problem I guess. I will look into this.