[Asterisk-Users] Problem with ADIT 600 and FXO configuration

[William K. Volkman]

I've looked through the archives of the mailing list for the
last year and although informative I've not been successful
at get this to work.  We had a working Asterisk PBX system
with 3 Digium X101P FXO lines and two TDM400P FXS cards.
I've setup an ADIT 600 with an 8 port FXO card (and an
8 port FXS card not currently installed).  We are going
to be adding a T1 for incoming calls this week. I removed
two of the X101P cards and installed a TE406P.  I'm using
Asterisk 1.0.9 (and matching zaptel, libpri) from tar files.

/etc/zaptel.conf has this configuration:
span=1,1,0,esf,b8zs,yellow
span=2,0,0,esf,b8zs
span=3,0,0,esf,b8zs
span=4,0,0,esf,b8zs
#Modular unit, first card is FXO
fxsks=1-3
unused=4-8
#Modular unit, 1 FXS cards
unused=9-16
unused=17-24
unused=25-48,49-72,73-96
fxsks=97
fxoks=98-101
fxoks=102-105

/etc/asterisk/zapata.conf has this:
group = 0
signalling=fxs_ks
context = incoming
busydetect = yes
overlapdial = no
channel => 1-3

signalling=fxs_ks
channel => 97  ;X100P 

group = 1
signalling = fxo_ks
context = internal
;TDM400P
callerid = "Available" <200>
channel => 98-100
callerid = "xxxxx"
channel => 101
;TDM400P
callerid = "xxxxx"
channel => 102
callerid = "xxxxx"
channel => 103

Parts of my adit configuration:
-Setting slot a.
                 
set a:1 up
set a:1 fdl none
set a:1 lbo 4
set a:1 framing esf
set a:1 id "Inbound"
set a:1 linecode b8zs
set a:1 loopdetect csu
set a:1:1-24 side drop
set a:1:1-24 type voice
set a:1:1-24 signal ls
set a:2 up
set a:2 fdl none
set a:2 lbo 1
set a:2 framing esf
set a:2 id "Outbound PBX"
set a:2 linecode b8zs
set a:2 loopdetect csu
set a:2:1-24 side drop
set a:2:1-24 type voice
set a:2:1-24 signal ls
                       
-Setting slot 1.
                 
set 1:1-8 signal lscpd
set 1:1-8 txgain -3
set 1:1-8 rxgain -6
                    
-Setting primary and secondary clock sources.
                                              
set clock1 a:1
set clock2 internal
                    
-Setting the system idle pattern for DS0s.
                                           
set idle 0xff
              
-Making connections.
                     
connect a:2:1-3 1:1-3
                      
Inbound calls just ring and ring (the leds on the ADIT change
state) however asterisk doesn't respond.  Attempts to make
outgoing calls get:
    -- Executing Dial("SIP/202-ba07", "Zap/g0/5551212") in new stack
Nov 28 02:54:45 NOTICE[8627]: app_dial.c:764 dial_exec: Unable to create
channel of type 'Zap'
  == Everyone is busy/congested at this time
    -- Executing Congestion("SIP/202-ba07", "") in new stack
  == Spawn extension (from-sip, 95942060, 3) exited non-zero on
'SIP/202-ba07'
    -- Executing Hangup("SIP/202-ba07", "") in new stack

I've tried just about all combinations of gs/ls/ks for the
signalling to no avail.  Here is the output of status:

> status a:2:1-3
                 
DS0         Rx AB  Tx AB  Signal  T1                 TP
---         -----  -----  ------  -----------------  --
a:2:1         01     01      LS   Traffic            N
a:2:2         01     01      LS   Traffic            N
a:2:3         01     01      LS   Traffic            N

> status 1:1-3
               
FXO    Rx AB  Tx AB  Signal=>T1 Sig  T1                 TP
---    -----  -----  --------------  -----------------  --
1:1      01     01   LSCPD => LS     Traffic             N
1:2      01     01   LSCPD => LS     Traffic             N
1:3      01     01   LSCPD => LS     Traffic             N
                                                           
> show connect a:2:1-3
    From            Desc                    Desc            To
 -----------  ------------------      -----------------  ---------
  A:02:01        LS VOICE   DS0 <--> FXO    VOICE LSCPD   1:01
  A:02:02        LS VOICE   DS0 <--> FXO    VOICE LSCPD   1:02
  A:02:03        LS VOICE   DS0 <--> FXO    VOICE LSCPD   1:03

Can anyone spot what I've got wrong?  Any suggestions or hints
welcome.

Thanks,
William.