Unity Express - Incoming calls wont get voice mail

CUE works fine with telephones on my local network. Incoming and outgoing calls work fine.
However when I get an incoming call via SIP trunk the call will not get forwarded to unity express after 10 seconds. The line goes dead.
I searched for another post which suggested the following commands:
telephony-service
call-forward pattern .T
voice service voip
allow connections from h323 to sip
I've double checked them and there's still something wrong.
Here's my current configuration:
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
h323
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
telephony-service
load 7910 P00403020214
load 7960-7940 P00305000301
max-ephones 24
max-dn 24
ip source-address 192.168.20.1 port 2000
auto assign 1 to 24
system message Comtek
voicemail 3000
max-conferences 8 gain -6
call-forward pattern .T
moh music-on-hold.au
time-webedit
transfer-system full-consult
transfer-pattern 2...
transfer-pattern 3...
directory last-name-first
directory entry 2 2001 name Phone Two 7912
directory entry 3 2000 name Phone One 7970
ephone-dn 1 dual-line
number 2000 secondary 441833000000
call-forward busy 3000
call-forward noan 3000 timeout 10
no huntstop
ephone 1
no multicast-moh
device-security-mode none
mac-address 0017.0EF0.3642
type 7970
button 1:1
So pros, any suggestions?
Thanks

I made a new dial-peer to handle incoming calls as follows.
dial-peer voice 1000 voip
description Incoming SIP
translation-profile incoming SIPin
voice-class codec 1
session protocol sipv2
incoming called-number .T
dtmf-relay rtp-nte
no vad
The translation-profile puts the call through to my 2000 extension.
This is my "show call active voice brief" when an external incoming call is ringing through to my 2000 ephone-dn.
To me this seems to show the dial-peer "1000" matching and using the g711ulaw codec
Telephony call-legs: 1
SIP call-legs: 1
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
1715 : 552 596706500ms.1 +-1 pid:1000 Answer +441833696807 connecting
dur 00:00:00 tx:0/0 rx:0/0
IP 87.127.240.98:16188 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
1715 : 553 596706510ms.1 +-1 pid:20001 Originate 2000 connecting
dur 00:00:00 tx:0/0 rx:0/0
Tele 50/0/1 (553) [50/0/1.0] tx:0/0/0ms None noise:0 acom:0 i/0:0/0 dBm
Telephony call-legs: 1
SIP call-legs: 1
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
This is the "show call active voice brief" for an external incoming call when the call is established.
Telephony call-legs: 1
SIP call-legs: 1
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
1731 : 569 597220040ms.1 +3730 pid:1000 Answer +441833696807 active
dur 00:00:02 tx:105/16800 rx:104/16640
IP 87.127.240.98:15162 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
1731 : 570 597220060ms.1 +3700 pid:20001 Originate 2000 active
dur 00:00:02 tx:0/0 rx:105/16800
Tele 50/0/1 (570) [50/0/1.0] tx:16180/16180/0ms g711ulaw noise:0 acom:0 i/0:0/0 dBm
Telephony call-legs: 1
SIP call-legs: 1
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
Not too sure where to go from here.

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http://www.cisco.com/en/US/products/sw/voicesw/ps2237/products_configuration_example09186a008015ceec.shtml#steps
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