07-31-2008 07:46 AM - edited 03-15-2019 12:20 PM
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.
07-31-2008 12:49 PM
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.
07-31-2008 03:43 PM
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.
07-31-2008 03:57 PM
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
07-31-2008 06:22 PM
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).
08-01-2008 08:15 AM
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
08-01-2008 08:18 AM
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.
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