CIsco CME not redirecting inbound PSTN VOIP to CUE voicemail

I am setting up a small office environment on a Cisco 2811 Router running advipservices 12.4 (15)T with CME 4.1(0) and a AIM-CUE running CUE 3.2.3. I have three 7960G IP phones running SCCP one of which is the receptionists which has three lines configured, one Internal and two external #1 (DDI 1) external #2 (DDI2) which correspond to ephone-dn #1, 8 & 9. The other 2 phones have internal extensions only.

ephone-dn 8 and 9 have configurations like :-

ephone-dn 8 dual-line
 number 01234xxxxxx secondary 2884
.
.
call-forward busy 8888
call-forward noan 8888 timeout 10

I have a dial-peer setup for 8888 to point to the CUE

On CUE I have created users and mailboxes for the three users and two GDM, which i am using as the receptionist has to have access to the other two user mailboxes. This all appears to work, All three users can access their personal mailbox, and can access the GDMs that they are a member of.

What I am have issue with is the transferring of calls to voicemail. If i call 2884, or 2885 internally (the secondary numbers on line 2 and 3 of the reception phone), I get redirected to the correct GDM for that extention. If I call the DDI number associated with line 2 and 3 on the reception phone, Again from an internal voip phone, I get redirected to the correct GDM, and I can see that the system is calling 8888 (voicemail number) on the internal VoIP phone i'm call ing from. However, if I make an inbound external call to the DDI number, the following happens:-

reception phone rings on correct line for 10 seconds
reception phone stops ringing and but screen does not show that it's redirecting to 8888
on external phone ringing tone is still heard.

I have run some debug on the CME router, and i can see the call coming in OK, and also that it gets redirected to 8888 with reason=2, but for some reason it is not being picked up by CUE or redirected to a mailbox.

I've obviously missed a step in the setup, but for the life of me I cannot see it.
LVL 3
ElvorfinAsked:
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sr75Commented:
Can you attach a show run | sec dial-peer output?
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ElvorfinAuthor Commented:
Yup here it is:

MWM-Cisco#sh run | sec dial-peer
dial-peer voice 2 voip
 description Outgoing SIP trunk to 01 Landline
 translation-profile outgoing PSTN
 destination-pattern 01.........
 voice-class codec 1
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 no vad
dial-peer voice 3 voip
 description Outgoing SIP trunk to 02 Landline
 translation-profile outgoing PSTN
 destination-pattern 02.........
 voice-class codec 1
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 no vad
dial-peer voice 4 voip
 description Outgoing SIP trunk to 03 Landline
 translation-profile outgoing PSTN
 destination-pattern 03.........
 voice-class codec 1
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 no vad
dial-peer voice 5 voip
 description Outgoing SIP to mobiles
 translation-profile outgoing PSTN
 destination-pattern 07.........
 voice-class codec 1
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 no vad
dial-peer voice 800 voip
 description Outgoing 0800
 translation-profile outgoing PSTN
 destination-pattern 0800T
 voice-class codec 1
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 no vad
dial-peer voice 845 voip
 description Outgoing 0845
 translation-profile outgoing PSTN
 destination-pattern 0845.......
 voice-class codec 1
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 no vad
dial-peer voice 870 voip
 description Outgoing 0870
 translation-profile outgoing PSTN
 destination-pattern 0870.......
 voice-class codec 1
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 no vad
dial-peer voice 844 voip
 description Outgoing 0844 numbers
 translation-profile outgoing PSTN
 destination-pattern 0844.......
 voice-class codec 1
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
dial-peer voice 966 voip
 description Saudi Arabia
 translation-profile outgoing PSTN
 destination-pattern 00966.........
 voice-class codec 1
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
dial-peer voice 8888 voip
 description Voicemail
 translation-profile outgoing TO_CUE
 destination-pattern 8888
 session protocol sipv2
 session target ipv4:192.168.1.199
 dtmf-relay rtp-nte
 codec g711ulaw
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sr75Commented:
What are you translating with this command?

 translation-profile outgoing TO_CUE
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ElvorfinAuthor Commented:
The number translation is as follows, (with real phone numbers hidden):

voice translation-rule 2
 rule 1 /01xxxxxx884/ /2884/
 rule 2 /01yyyyyy885/ /2885/
!
!
!
voice translation-profile TO_CUE
 translate called 2
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ElvorfinAuthor Commented:
Anybody got any ideas about this please?
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ElvorfinAuthor Commented:
I've solved this and it's an issue with the SIP trunks not being concatenated.
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ElvorfinAuthor Commented:
The problem has evolved since i raised this question and the issue described is no longer valid.
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