[Asterisk-Users] TDM400P Internal Extenion Config

[Nick Cobley]

Hopefully someone here can save my sanity. I have been trying to solve 
this problem for days now, but just cant put my finger on it. Im new to 
* so I have probably done something stupid!

I have a TDM400P with one FXO module and a FXS module. The main problem 
I have is not being able to get the extension attached to the FXS module 
to ring or be able to make calls. It gets a dialtone fine but I guess 
this doesnt really mean all that much.

If I dial the extension I just get a 404 error on the phone 
(Grandstream), but no errors at all on the console. I am using 
CVS-HEAD-07/14/04.  Here is a snippet of what I have in the various 
config files.


zaptel.conf

loadzone=au
defaultzone=au
fxsks=1
fxoks=2

zapata.conf

[channels]
context=incoming
switchtype=national
signalling=fxs_ks
usecallerid=yes
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
txgain=0.0
rxgain=0.0
group=1
callgroup=1
pickupgroup=1-4
immediate=no
busydetect=yes
busycount=7

callerid=asreceived
channel => 1
context=internal
group=2

signalling=fxo_ks
callerid="Fax" <310>

channel => 2


extensions.conf

[general]
static=yes
writeprotect=no

[globals]

[sip]
exten => 301,1,Dial(SIP/Nick,20,tr)
exten => 302,1,Dial(SIP/Sharon,20,tr)
exten => 1000,1,Dial(SIP/Nick&SIP/Sharon,20,tr)
exten => 302,2,VoiceMail,u302
exten => 301,2,VoiceMail,u301
exten => 1000,2,VoiceMail,u9999
exten => 1000,102,VoiceMail,b9999
exten => 1001,1,Ringing
exten => 1001,2,Wait(2)
exten => 1001,3,VoicemailMain
include => outgoing

[incoming]
exten => s,1,Dial(SIP/Nick&SIP/Sharon,20,tr)

[outgoing]
exten => _7.,1,Dial(IAX2/login:passwd at XXX.XXX.XXX>XXX/${EXTEN:1})
exten => 5.,1,Dial,Zap/1/${EXTEN:1}

[9103]
exten => 21060,1,Dial(SIP/Nick)
exten => 21062,1,Dial(SIP/Sharon)

[internal]
exten => 310,1,Dial,Zap/2



If I try to make any calls from the extension connected to the fxs 
module i just get what sounds like a busy tone. Looking at the console 
it generally give the error "zt_set_hook: zt hook failed Device or 
resource busy". It only gives this error when it goes off hook and 
number dialed.

Only other information I can provide is a couple errors when asterisk 
start up. I get the following:

"Unable to open /dev/dsp: No such device"
"Unable to get our IP address, Skinny disable"
"Ignoring switchtype"

Have not been able to dig out vast amounts of information on the above, 
but what I have found didnt seem to point to my problem, but then what 
do I know!

If anyone can help I would appreciate it! I'm going crazy here!

Kind regards
Nick