Cisco Unified CME in SRST Fallback Mode with CUE

wbarren · ‎07-31-2008

Is it possible to have Cisco Unified CME in SRST Fallback Mode setup and when phone fallback use CUE as their voicemail? During normal operations the voicemail will go to unity.

Kenneth Mohammed · ‎07-31-2008

I know you can use CME in SRST fallback as per this document:

http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmesrst.html

As far as CUE is concerned, I don't see a reason why it wouldnt work as long as you have the correct configuration (ephone-dn's,call-forward noan/call-forward busy, dial-peers, and telephony-service) pointing to the CUE for voicemail when in SRST mode. However, I have been having trouble finding a document to support this theory.

Hope that helped.

maharris · ‎07-31-2008

This tech note is the only place I could find that actually discusses it, and describes how to do it, it does appear to be supported, I don't know that we have tried it yet:

Configure Cisco SRST (Optional)

Cisco SRST is used for emergency phone and voice mail services when the WAN that connects a remote site to a Cisco CallManager is down. There is nothing to do if the Initialization Wizard is used to set up the system in Cisco Unity Express. You can use this basic configuration in the Cisco IOS configuration:

dial-peer voice 1 voip

description Local NM-CUE (CME) Voicemail

destination-pattern 28000

session protocol sipv2

session target ipv4:172.18.106.107

dtmf-relay sip-notify

codec g711ulaw

no vad

!

dial-peer voice 2 voip

description Local NM-CUE (CME) Auto Attendant

destination-pattern 28100

session protocol sipv2

session target ipv4:172.18.106.107

dtmf-relay sip-notify

codec g711ulaw

no vad

!

dial-peer voice 3 voip

description Local NM-CUE (CME) Greeting Management System

destination-pattern 28111

session protocol sipv2

session target ipv4:172.18.106.107

dtmf-relay sip-notify

codec g711ulaw

no vad

!

call-manager-fallback

ip source-address 172.18.106.105 port 2000

max-ephones 52

max-dn 208

voicemail 28000

call-forward busy 28000

call-forward noan 28000 timeout 12

!

The three configured destination patterns (28000, 28100, and 2111) correspond to the three DNs assigned to the route points. The CTI ports are not referenced anywhere.

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

mattcalderon · ‎07-31-2008

I found this when searching for some information and it touches on SIP support with CUE in SRST.

Realize this is for 3.2 and they are up to 4.1.

http://www.cisco.com/en/US/docs/ios/12_3t/12_3t11/feature/guide/srs_sip.html

wbarren · ‎07-31-2008

I spoke with our local Cisco Engineer and his response was the following. They both look like pretty good directions to go.

1: You can setup a dial-peer in the Branch router that points all voicemail requests out over the PSTN and back into the HQ location. It works great. (NOTE: This would obviously eliminate the need for a CUE module).

2: You can network the CUE modules with Unity anytime, keep the users on their local CUE for voicemail, and the event that connectivity is lost, they will not be without CUE functionality (i.e. ACD, AA and Voicemail).

maharris · ‎08-01-2008

Yes, the advantage of both of those is that they only need to keep up with one voicemail system. We usually see the 'dial out the PSTN' approach, if they need to access their voicemail. One consideration might be if they have their numbers coming in their own PRI at the remote location, or if all calls come into the main location and get distributed out to the remotes, things like that, specific characteristics of how they have the network designed.

Mary Beth

wbarren · ‎08-01-2008

In our environment each office has their own PRI which local numbers come in on. So for us I am starting to believe the best approach is going to be Networking Unity to Unity Express.