Call dropping after picking option in CUC

[rafaelrangel]

I have the following scenario: ITSP-CUBE-CUCM-CUC When the call comes in through CUBE, it uses a dial-peer to translate to a DN. This DN is associated with a CTI POINT in my CUCM that plays for a voicemail in the CUC.

I listen to GREETING and after choosing an option the connection is not transferred to the HUNT PILOT for that choice. She drops and the group phones do not ring.

By doing a test directly from a telephone I listen to GREETING, and after choosing it works perfectly.

The CALL HANDLER settings in CUC are correct by this test.

Can anyone tell if I'm missing out on some detail?

[Marco Rojas Abarca]

What type of connection do you have from CUBE to ITSP?
Also, is CUC integrated SCCP? or SIP?
Would be worth looking at, simultaneously, debugs and CCM Traces from the working vs. non-working to determine what is happening.

[rafaelrangel]

Hello Marco.

I have a Sip Trunk from my Cube with my ITSP.
Between my CUCM and my CUBE I have a Sip trunk too.
From my CUC to my CUCM I have SCCP.

Follow the dial-peers:

dial-peer voice 100 voip
description INBOUND - FLUX TRUNK TO CUCM
huntstop
preference 1
destination-pattern 3100
session protocol sipv2
session target ipv4: 192.168.200.2
dtmf-relay rtp-nte
do not go
!
dial-peer voice 101 voip
translation-profile incoming ITSP-IN-RJ
session protocol sipv2
session target sip-server
incoming called-number 212391XXXX
voice-class codec 1
dtmf-relay rtp-nte
!
dial-peer voice 102 voip
translation-profile outgoing ITSP-OUT-RJ
destination-pattern.
session protocol sipv2
session target ipv4: XXX.XXX..XXX..XX
voice-class codec 1
dtmf-relay rtp-nte
do not go
!
dial-peer voice 103 voip
destination-pattern.
session protocol sipv2
session target sip-server
incoming called-number 0T
voice-class codec 1
dtmf-relay rtp-nte
do not go
!