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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.

sr75

Can you attach a show run | sec dial-peer output?

Elvorfin

ASKER

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

sr75

What are you translating with this command?

translation-profile outgoing TO_CUE

Elvorfin

ASKER

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

Elvorfin

ASKER

Anybody got any ideas about this please?

ASKER CERTIFIED SOLUTION

Elvorfin

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Elvorfin

ASKER

The problem has evolved since i raised this question and the issue described is no longer valid.