Cisco Ip phone but troubles with AMbient MD3200 card

HI
Im a newbee and i’ve some trouble to make my cisco 7940 series work with asterisk
I want to make an internal network between my computer and IPphones connexted with a cisco 2900 series XL
the phone test as an ip adress given by my dhcp server and answer to the ping but when I type sip show user on aserisk Ive nothing

thx for answers

sip.conf
[general]
context=default ; Default context for incoming calls

bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)

disallow=all ; First disallow all codecs
allow=alaw
allow=ulaw ; Allow codecs in order of preference
allow=ilbc ;
canreinvite=yes
host=dynamic
;musicclass=default ; Sets the default music on hold class for all SIP calls
; This may also be set for individual users/peers
language=fr ; Default language setting for all users/peers

[10]
type=friend
context=10
username=10
fromuser=10
host=dynamic
callerid=<10>

[11]
type=friend
context=11
username=11
fromuser=11
host=dynamic
callerid=<11>

extentions.conf

[general]
STANDARDISTE=>SIP/10
DIRECTION=>SIP/11

[direction]
exten => 10,1,Dial(SIP/11,40,t)
exten => 10,2,Answer
exten => 10,3,Setlanguage(fr)
exten => 10,4,Hangup

[standard]
exten => 11,1,Dial(SIP/10,40,t)
exten => 11,2,Answer
exten => 11,3,Setlanguage(fr)
exten => 11,4,Hangup

the line 1 SIP phone configuration is :

name 10
Shortname 10
Authentication 10
pass vide
displayname 10
PROXY 192.168.3.100 (Ip du serveur asterisk)
PROXY PORT 5060

Does cisco support iLBC Codec ?

My ip phones work without any problem

I want to know if it’s possible with 2 Ambient md3200 cards bought on EBAY to insert 2 analogics phones into my IP phone network (the cards are Tiger3XX Modem/IDSN interface)

le ztcfg -vvvv doesn’t display any error

In fact I’ve configured zapata and zaptel but I dont have tonnality (it’s plugged to the first card on phone port
here are my files

Zaptel Configuration File

fxsks=1-2
loadzone=fr
defaultzone=fr

zapata conf file

[channels]
context=ligne1
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
rxgain=0.0
txgain=0.0
group=0
callgroup=1
pickupgroup=1
immediate=no
channel=>1

signalling=fxs_ks
channel=>1
context=ligne1

signalling=fxs_ks
channel=>2
context=ligne2
language=fr

extensions.conf

[general]
;

STANDARD=>SIP/10
DIRECTION=>SIP/11
DIALOUTANALOG=>Zap/1

[direction]
exten => 11,1,Dial(SIP/11)
exten => 10,1,Goto(standard,${EXTEN},1)
exten => 12,1,Goto(DIALOUTANALOG,${EXTEN},1)
;exten => 11,2,Answer()
;exten => 11,3,Setlanguage(fr)
;exten => 11,4,Hangup()

[standard]
exten => 10,1,Dial(SIP/10)
exten => 11,1,Goto(direction,${EXTEN},1)
exten => 12,1,Goto(DIALOUTANALOG,${EXTEN},1)
;exten => 12,1,Goto(ligne1,${EXTEN},1)
;exten => 13,1,Goto(ligne2,${EXTEN},1)
;exten => 10,2,Answer
;exten => 10,3,Setlanguage(fr)
;exten => 10,4,Hangup

[ligne1]
exten => 11,1,Dial(ZAP/1)
exten => 10,1,Goto(standard,${EXTEN},1)
exten => 12,1,Goto(DIALOUTANALOG,${EXTEN},1)