ISR G2 - cme-srst license

On ISRG2, in order to use cme-srst feature, just installing uck9 license is not good enough.  You also need to activate the cme-srst license by accepting the end user agreement.  After that you can use this feature for 12 years.

http://www.cisco.com/en/US/docs/routers/access/sw_activation/SA_on_ISR.html#wp1155517

Here is the command you need to issue:
config t
license accept end user agreement

reload your router then check:


SRST-R2#sh license det cme-srst
Feature: cme-srst                        Period left: 8  weeks 3  days
Index: 1        Feature: cme-srst                          Version: 1.0
        License Type: EvalRightToUse
        License State: Active, In Use
            Evaluation total period: 8  weeks 4  days
            Evaluation period left: 8  weeks 3  days
            Period used: 38 minutes 3  seconds
            Transition date: Jan 21 2013 17:12:02
        License Count: 20/0  (In-use/Violation)
        License Priority: Low
        Store Index: 7
        Store Name: Built-In License Storage

Be careful that before you do this step, you can still issue the CME-SRST command, it just doesn't work!  The phone cannot be registered with the reject message

Nov 22 09:08:32.987: %IPPHONE-6-REG_ALARM: 24: Name=SEP588D09357799 Load=SCCP6945.9-3-1-3 Last=Phone-Reg-Rej

My IOS version:  15.2(4)M2

MGCP Gateway Fallback Configuration Example

When you are deploying MGCP in branch office and the WAN connection between the MGCP gateway and UCM is broken, you can enable fallback on IOS gateway and route call based on your H.323 configuration.

These are the commands that you need to configure on your MGCP gateway.

R1(config)#ccm-manager fallback-mgcp

R1(config)#application

R1(config-app)#global

R1(config-app-global)#service alternate Default

Also configure the necessary incoming and outgoing voip and pots dial-peer as well as SRST configuration to handle the fallback situation.

Cisco UCM - Voice Mail Box Mask on Voice Mail Profile

The mask only mask the ReDir number.  Not the caller number, therefore:

When branch phone 5002 calls 5001 redir to 1220 (VM Pilot number, CUC in hub site) in SRST mode, it will mask the redir number from 98765001 (DID number) to 5001, therefore the caller 5002 can hear 5001's greeting instead of a general mailbox greeting.

However when 5001 press the voicemail envelop button on phone, the CLID is 98765001 therefore user 5001 doesn't able to listen to his personal greeting, as CUC doesn't recognize a subscriber has 98765001. It is not handled by the voice mail box mask.

One of the way is to enter 98765001 as alternate extension for the subscriber 5001.  If alternate extension is not a solution, you can use calling party transformation on UCM to manipulate the calling number, say for example create a new device pool for the voice mail ports on UCM and apply the calling party transformation CSS to this new device pool.

Cisco UCM + CUE integration - in SRST scenario

In previous post I have highlighted the steps and procedures that are required for a UCM + CUE JTAPI integration:

http://pandaeatsbamboo.blogspot.com/2012/02/ucm-cue-integration-example-step-by.html

What if the CUE is located at a branch site and the connection between UCM and CUE is broken?  When the WAN link is downed, SRST kicks in and the CUE can change from a JTAPI integration into a SIP integration, so that the SRST router can talk to CUE via SIP.

0. Assume you have the JTAPI integration in place, for normal situation when the connection between hub site and branch is up:
http://pandaeatsbamboo.blogspot.com/2012/02/ucm-cue-integration-example-step-by.html

1. Configure CUE SIP subsystem, the gateway address is your SRST router address:

ccn subsystem sip

gateway address "192.168.166.254"

mwi sip unsolicited

2. Configure SIP trigger, the voicemail pilot number:

ccn trigger sip phonenumber 1110

application "voicemail"

3. Configure the SIP dial-peer on your router

dial-peer voice 1110 voip

destination-pattern ^1110$

session protocol sipv2

session target ipv4:192.168.166.254

dtmf-relay sip-notify

codec g711ulaw

4. Configure the SIP MWI:

sip-ua

mwi-server ipv4:142.1.66.253 expires 3600 port 5060 transport udp unsolicited

5.  Sample CME as SRST configuration

telephony-service

srst mode auto-provision none

srst ephone template 1

srst dn template 1

srst dn line-mode octo

max-ephones 20

max-dn 20 no-reg

ip source-address 192.168.166.254 port 2000

voicemail 1110

mwi relay

max-conferences 8 gain -6

transfer-system full-consult

create cnf-files version-stamp 7960 Feb 25 2012 01:03:36

!

!

ephone-dn-template  1

call-forward busy 1110

call-forward noan 1110 timeout 20

mwi sip

huntstop channel 1

!

When the WAN link is downed, your branch phone can still access your local CUE, leave and retrieve voicemail as usual and the end user experience, MWI, etc will be the same. 

CUCME-SRST Cheatsheet

Starting from CUCME 4.0, it provides fallback functionality to IP phones that are registered to CUCM, just like SRST, but provide more robust features like B-ACD during the fallback mode.

Some of the key points for CME-SRST mode:

• First supported with CME 4.0 – IOS 12.4(4)XC
• IP phones re-home to CME in event of CallManager failure & have access to advanced CME telephony features
• Support for up to 240 ephones
• No support for Vg248 registration during fallback
• Lack of support for “alias” command
• Support for Cisco Unity at remote sites (Distributed Exchange/Domino)
• Pickup Groups, Hunt Groups, B-ACD, Call Park,Softkey templates, paging
• Support for CIPC 2.0 with VTA 2.0 on same computer
• No support for secure voice in SRST mode
• Requires more complex configuration

An iPhone app for CME Config Generation:
http://itunes.apple.com/us/app/cme-config-generator/id453025819?ls=1&mt=8

The call-manager-fallback command, which is used to configure Cisco Unified SRST, cannot be used on a router that is configured for Cisco Unified CME

Sample Configuration:
telephony-service
srst mode auto-provision all
srst ephone template 1
srst ephone description CCME-SRST MODE : Oct 31 2006 19:00:20 : Nov 01 2006 00:31:17
srst dn template 1
srst dn line-mode dual
max-ephones 144
max-dn 288
ip source-address 1.4.19.1 port 2000
max-conferences 8 gain -6
transfer-system full-consult
create cnf-files version-stamp Jan 01 2002 00:00:00