Newbie, Can't get my TDM400P to respond?

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.

what type of respond do you want?

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?

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.

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.

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.