[Problem] Digium TDM400P - Can't dial Extensions

I have just gotten my asterisk system installed and set up, compiling from source (1.2.5) on a SUSE 9.1 system. When I turn on my phone, I hear a dialtone. As soon as I dial, I get a rapid beep, like I dialed an invalid number. I get the following errors from the CLI in Verbose 10
– Starting simple switch on ‘Zap/2-1’
– Hungup 'Zap/2-1’
Mar 10 21:45:06 WARNING[6413]: chan_zap.c:1586 zt_set_hook: zt hook failed: Device or resource busy

I also tried Channel one with the same results.
zttool and ztcfg show the hardware to be working properly



;define any trunk groups

; hardware channels

; define channels
channel => 3

channel => 2

exten => 611,1,Answer()
exten => 611,2,Echo()
exten => 613,1,Dial(IAX2/iaxfwd/613)

exten => 754946,1,Answer()
exten => 754946,2,Echo()

Any Help would be much appriciated. Please remember that it has been several years since I played with linux and I wasn’t all that good to begin with.

If you’re really that much of a beginner, I’d really recommend using asterisk@home…

But then I would remain a beginner forever…I have plenty of time and a renewed need to learn linux…this seems like as good a time as any to learn. I didn’t know much about mechanics until I bought an old CJ-7 and rebuilt it.

not clear on your set up. what kind of modules do you have in the TDM400 and what kind of phone are you using, analog?

that’s just silly. when you’re starting with something, you take a graduate level course? you can’t move up as you learn? as far as your question in general, i see your point, but why should we help someone who IS a rank beginniner but wants to start on 3rd base?

I have a TDM400P with two FXS modules (1,2) and two FXO modules (3,4). I have an analog phone connected to channel 2, though i have tried channel 1 with the correct changes to my config files with the same results. I pickup the phone and hear a dialtone. I dial the extension from my extensions.conf (I’ve tried different extension numbers) and then get the steady beeping sound through the phone. I also get the error messages I posted above from my CLI console. This is only a test environment and I don’t currently have an incoming Pots line to test the FXO modules on. Thanks!!!

Without getting into a long discussions of my motives, I feel I learn better by taking on large projects and working my way through them. I only ask questions when I have been stumped for a long time. I appriciate the advice on switching to asterisk@home, based on my personal situation I have chosen to remain on this course. If you choose to give me no more advice, I understand completely.

have you tried channels 3 & 4? i mean are you sure your settings in zaptel.conf match your card? maybe the FXO & FXS modules are swapped? have you read the book in my signature for clues on proper configuration? i have a TDM400 with one FXO module which worked off-the-bat so don’t have much too experience troubleshooting this error.

when i decided to learn Asterisk I jumped right in and skipped AAH. i think that if you’ve got the time it’s a good way to do it. i am now using AAH and am glad that i know what is going on ‘under the hood’.