asterisk users - Nov 2006 - Need help connecting Alcatel 4400 PBX to Asterisk

Hi there

I have a TE110P card fitted in my linux box running :
Linux version 2.6.9-5.ELsmp (bhcompile@decompose.build.redhat.com) (gcc version
3.4.3 20041212 (Red Hat 3.4.3-9.EL4)) #1 SMP Wed Jan 5 19:30:39 EST 2005

I followed the installation steps on digium website...no errors reported.
The modules seem to have loaded...here's what lsmod shows:
Module                  Size  Used by
wcte11xp               30496  31
zaptel                196740  67 wcte11xp

still the light on my card is off....does that mean the card has not initialised
properly?

On loading Asterisk, I do not get any errors, but I do see these warnings:
Parsing '/etc/asterisk/zapata.conf': Found
Nov  1 11:57:21 WARNING[3454]: chan_zap.c:10874 setup_zap: Ignoring switchtype
Nov  1 11:57:21 WARNING[3454]: chan_zap.c:10874 setup_zap: Ignoring signalling

on running asterisk -cvvv . I do see Aterisk Ready at the end ...then What do
these warnings mean?

Also, I DO NOT get these lines on asterisk startup:-
channel 0/1 successfully restarted on span 1
    -- B-channel 0/2 successfully restarted on span 1
    -- B-channel 0/3 successfully restarted on span 1
    -- B-channel 0/4 successfully restarted on span 1
    -- B-channel 0/5 successfully restarted on span 1
    -- B-channel 0/6 successfully restarted on span 1

does that mean my channels are not available?

*CLI> zap show status
Description                              Alarms     IRQ        bpviol     CRC4
Digium Wildcard TE110P T1/E1 Card 0      OK         0          0          0

*CLI> pri show span 1
Primary D-channel: 16
Status: Provisioned, Down, Active
Switchtype: EuroISDN
Type: CPE
Window Length: 0/7
Sentrej: 0
SolicitFbit: 0
Retrans: 0
Busy: 0
Overlap Dial: 0
T200 Timer: 1000
T203 Timer: 10000
T305 Timer: 30000
T308 Timer: 4000
T313 Timer: 4000
N200 Counter: 3

---------------------------
here's my extensions.conf:
[general]
static=yes
writeprotect=no

autofallthrough=yes

[sip]
exten => 9820,1,Dial(SIP/iyer)
exten => 9821,1,Dial(SIP/shweta)
exten => 9810,1,Dial(SIP/shashi)
exten => 9851,1,Dial(Zap/g1/851,20)
[incoming]
exten => s,1,Answer()
exten => s,2,Playback(hello-world)
exten => s,3,Hangup()
exten => 9821,1,Dial(SIP/shashi)
exten => 9851,n,Dial(Zap/g1/851)
---------------------------

here's zapata.conf
[trunkgroups]
trunkgroup => 1,16
spanmap =>1,1,1

[channels]
switchtype=euroisdn
signalling=pri_cpe
context=incoming
language=uk
group=1
callgroup=1
pickupgroup=1
echocancel=yes
immediate=no
channel => 1-15,17-31


usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancelwhenbridged=yes

musiconhold=default
-------------------------------

here's zaptel.conf:

span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16

loadzone = us
defaultzone=us


---------------------------

Now the problem

I can call and talk SIP to SIP...here's what I see on asterisk CLI

-- Executing Dial("SIP/iyer-09326480", "SIP/shweta") in new
stack
    -- Called shweta
    -- SIP/shweta-0932b9c0 is ringing

But when I call zap extension, here's what I get:
 Executing Dial("SIP/iyer-09326480", "Zap/g1/851|20") in new
stack
Nov  1 12:07:55 NOTICE[3513]: app_dial.c:1056 dial_exec_full: Unable to create
channel of type 'Zap' (cause 34 - Circuit/channel congestion)
  == Everyone is busy/congested at this time (1:0/1/0)
  == Auto fallthrough, channel 'SIP/iyer-09326480' status is
'CONGESTION'

I have connected the PBX to digium card with the specified cable and done the
settings in PBX specified at:
http://www.voip-info.org/wiki/view/Alcatel+4400+via+PRI

What am I doing wrong?

I'd like to mention that on the Alcatel PX rack on the PRA2 card, the
NO-SIGNAL (NOS) light comes on when I shut down my linux box but it's off
when I load zaptel....doesn't that mean that PBX is able to sync to my
asterisk server?

Any help would be greatly appreciated.

Thanks in advance....

Kind Regards
Shweta


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