asterisk users - Sep 2004 - Digits being dropping when dialing from certain analog phones

FC2, Asterisk 1.0.0, Zaptel 1.0.0
TDM400P Port 1 FXS Port 4 FXO

Standard analogue handset plugged in with pstn line.

Problem:

I have 2 analog phones that I use, when plugged directly into pstn line
both phones work perfectly, dialing no issues. When I plug the handsets
into the TDM400P, one works perfectly the other drops random numbers.
Its like the tone is slightly different on the second handset and its
not picking up some numbers (1235&6 it seems). Is there a way to adjust
the tone detection, make it more sensitive?

Keys dialed from handset were

9 0418800185

I tried hitting the keys slowly as well as at my normal speed, all tones
are heard in the handset for all numbers.

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

Error in asterisk -vvvgc

    -- Starting simple switch on 'Zap/1-1'
    -- Hungup 'Zap/1-1'
    -- Starting simple switch on 'Zap/1-1'
    -- Executing Dial("Zap/1-1", "Zap/g2/088008") in new
stack
    -- Called g2/088008
    -- Zap/4-1 answered Zap/1-1
    -- Attempting native bridge of Zap/1-1 and Zap/4-1
    -- Hungup 'Zap/4-1'
  == Spawn extension (internal, 9088008, 1) exited non-zero on 'Zap/1-1'
    -- Hungup 'Zap/1-1'
    -- Starting simple switch on 'Zap/1-1'
    -- Executing Dial("Zap/1-1", "Zap/g2/0488008") in new
stack
    -- Called g2/0488008
    -- Zap/4-1 answered Zap/1-1
    -- Attempting native bridge of Zap/1-1 and Zap/4-1
    -- Hungup 'Zap/4-1'
  == Spawn extension (internal, 90488008, 1) exited non-zero on
'Zap/1-1'
    -- Hungup 'Zap/1-1'

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

/etc/zaptel.conf

fxols=1
fxsls=4
Loadzone=au

/etc/zapata.conf

[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
signalling=fxo_ls
callgroup=1
pickupgroup=1
immediate=no
context=internal
busydetect=yes
callerid="James Bean<690>"      ;assuming extension 690
mailbox=690                     ;stutter tone for voicemail - you can
use an optional context here
transfer=yes
channel=>1
group=2
signalling=fxs_ls
context=pstn
channel=>4

/etc/asterisk/extensions.conf

[pstn]

exten => s,1,NoOp(Comment Only: Call from ${CALLERIDNUM}) ; Just put a
comment in the CLI for info.
exten => s,2,Dial(Zap/g1,45,t)  ;Dial the group=1 zap card mod above
exten => s,3,Hangup

[internal]
exten => i,1,Playback(invalid)
exten => i,2,Hangup
exten => t,1,Hangup

exten => 099,1,Echo     ;simple echo test when you dial 099 on your
phone

exten => _9X.,1,Dial(Zap/g2/${EXTEN:1})
exten => _9X.,2,Congestion()