cancel
Showing results for 
Search instead for 
Did you mean: 
cancel
2105
Views
0
Helpful
3
Replies

CUE and SRST

kgroves42
Level 3
Level 3

I have a remote site with a CUE that is the primary Voice Mail. That seems to be working ok.  However when the site fails over to srst. We have two issues.

1- when you call another extension and it forwards to voicemail it does not go to the forwarded phones mailbox instaed it tries to log into the mailbox and gives the "Please enter your Pin" . However in normal mode it works fine.

2- We have POTS line that ring to an AA setup on the CUE, In SRST mode it does not allow a user to call an extension from the AA. Instead of transferring the Call it goes right to the users Voice Mail.

Any suggestions would be appreciated.

Thanks

Ken

3 Replies 3

Udit Mehrotra
Cisco Employee
Cisco Employee

Ken,

Is the CUE integrated with CUCM? If so, is this JTAPI integration?

If this is JTAPI integration then I do not think that this wil work, since when in SRST CUE needs to be integrated as SIP. Further thoughts are welcomed in this regards.

You may also wish to check - http://www.cisco.com/en/US/docs/voice_ip_comm/unity_exp/design/design21/cuemodel.html

Suggestions are welcomed.

--

Udit

Thanks for the response.

I am using UCM as my call control, and Jtapi as the CUE connection to UCM. I just followed this DOC:

http://www.cisco.com/en/US/partner/products/sw/voicesw/ps5520/products_configuration_example09186a0080289ef0.shtml

So you are saying if we switch to SIP it should start working? Is there a good Doc on how to set that up?

Thanks

Ken

Ken,

CUE can run either in CUCM or CCME mode. This depends on the license that is installed on CUE. This also decides if the integration would be JTAPI or SIP.

Since the CUE is integrated with CUCM, you need CCM licenses and hense this is JTAPI integration.Now with JTAPI would not work with SRST mode.

If you check, http://www.cisco.com/en/US/partner/products/sw/voicesw/ps5520/products_configuration_example09186a0080289ef0.shtml#srst

you can configure SIP dial-peers on SRST router which should ideally send the call to CUE and you should not lose your VM and AA. Along with dial-peers you will have to configure voicemail and CFNA and CFB under call-manager-fallback as shown in above link.

You will also have to create SIP triggers within CUE as show in the guide.

ASFAIK, when in SRST, MWI would not work for the IP Phones.

HTH.


~~

Udit

Getting Started

Find answers to your questions by entering keywords or phrases in the Search bar above. New here? Use these resources to familiarize yourself with the community: