​On CUCM
(CUCM Version Used for Screenshots: 11.5.1.11900-26)
System --> Security --> Sip Trunk Security Profile
Copy the ‘Non Secure SIP Trunk Profile’, enter a new name: For example: ‘CUC SIP Security Profile’. Leave everything else to default.

Advanced Features --> Voicemail --> Voice Mail Pilot
Either change the ‘Default’ one or create a new one.

​(We are not adding any Calling Search Space – but it has to be added if needed)
Advanced Features --> Voicemail --> Voice Mail Profile
Now again, either change the ‘Default’ one or create a new one.

​Device --> Trunk --> Add New

Type in a recognizable name, and add other parameters through the dropdown (like Device Pool, Location, etc)

​Leave other options as default, but make sure you select the following two:
‘Redirecting Diversion Header Delivery – Inbound’ in the ‘Inbound Calls’ section and

​‘Redirecting Diversion Header Delivery – Outbound’ in the ‘Outbound Calls’ section

Put the IP Address of the Unity Connection Server in ‘Destination Address’, select ‘SIP Trunk Security Profile’ which was created in the previous section.
SIP Profile is a required field, you can select ‘Standard SIP Profile’ for this purpose. Save the trunk configuration.

​Call Routing --> Route/Hunt --> Route Pattern --> Add New
Put in the Voice Mail Pilot DN here as the Route Pattern and select the SIP Trunk created in the previous step.

​(Keep in mind, I have not created, ‘Route Group’, ‘Route List’ and then applied it here. That method is fine too)
At this time, CUCM side of configuration is complete. However, make sure the phones are configured for the phone Voice Mail profile which was created. And they are configured to send un-answered calls to the voice mail.

On CUC
(CUC Version Used for Screenshots: 11.5.1.11900-26)
Telephony Integrations --> Phone System
Add a new one or select the existing one

​Click on ‘Go’ beside ‘Add Port Group’

​Select ‘SIP’ as the ‘Port Group Type’, ‘Display Name’ would be pre-populated, enter the IP Address of the CUCM Server in ‘IPV4 Address of Host Name’ and Save.

​Click on ‘Register with SIP Server’ leave the others to default settings.
You can also change the codec advertising if needed.

​Click on ‘Go’ beside ‘Add Ports’.
Note 1: SIP Integration of CUC and CUCM doesn't have MWI On/Off DNs. SIP Integration uses SIP NOTIFY messages to turn ON/OFF the MWI on the phone. 
Note 2: Ports dictate how many concurrent sessions can occur between CUCM and CUC. In this example, maximum of 3 sessions would exist between CUCM and CUC (for example 3 people would be able to check voicemail at one time because we are using 3 ports or 2 people can check voice mail and one auto attendant session, or any other combination) 
PS: Until CUC version 8, number of ports was dictated by license. From CUC 9 and later, this is no longer a licensed feature. Ports now depend on the hardware/hardware OVA template used.
For CUC 11.x, check this URL for the maximum number of ports supported:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/connection/11x/supported_platforms/b_11xcucspl.html#ID-2325-00000015

​Click on ‘Go’ beside ‘Check Telephony Configuration’ to check if everything is good.

​If no problems exist, you will see the following:

​Now the only thing left is to create voicemail boxes and you are good to go.

!Step 1 - CME CONFIG
!
interface SM1/0
 ip unnumbered Vlan2 ! Meaning we would be using IP address from the range of: Vlan2 range.
 service-module ip address 192.168.1.251 255.255.255.0
 !Application: CUE Running on SM
 service-module ip default-gateway 142.102.66.250 ! Default Gateway
!
ip route 142.102.66.253 255.255.255.255 SM1/0 ! Subnet mask is 255.255.255.255 as only traffic to this address 253 would come to Service Module 1/0
!
dial-peer voice 2222 voip !Dial-Peer for VM Pilot
 destination-pattern 2222$
 session protocol sipv2
 session target ipv4:192.168.1.251
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad


! ************
! For SCCP CME
! ************
telephony-service
 voicemail 2222
 mwi relay
!
 
ephone-dn  1  octo-line ! Put under existing DNs
 call-forward busy 2222
 call-forward noan 2222 timeout 20
 mwi sip
!
! ***********
! For SIP CME
! ***********
voice register global
 voicemail 2222
!
!
voice register dn  1 ! Put under existing DNs
 call-forward b2bua busy 2222  
 call-forward b2bua mailbox 2222
 call-forward b2bua noan 2222 timeout 20 
 mwi
!
voice register pool 1
 dtmf-relay rtp-nte ! Make sure DTMF Relay Method is already configured on the phone
!
sip-ua 
 mwi-server ipv4:192.168.1.251  unsolicited
!
!Step 2 - CUE CONFIG

!NOW THE CUE MODULE IS RECONGNISED BY THE CME AND ALL NECESSARY CONFIG IS COMPLETE
!REMEMBER CUE IS A SEPERATE MODULE RUNNING INSIDE THE CME ROUTER SO IT NEEDS TO BE INITIALISED TOO

service-module service-Engine 1/0 session

!Create a CCN Subsystem SIP

ccn subsystem sip
 gateway address 192.168.1.250
 mwi sip unsolicited
 dtmf-relay rtp-nte
 end subsystem
 
! Create a CCN Trigger SIP 

ccn trigger sip phonenumber 2222 !<-- This number has to match the number configured under "telephony-service" and under "voice register global"
 application voicemail
 enabled
 end trigger

!Create Users
 
username one create
username two create

username one phonenumber 2001
username two phonenumber 2002

username one pin 12345
username two pin 12345

voicemail mailbox owner one
no tutorial ! Optional - Incase you dont want the voicemail tutorial to be played for the first time users logs into Unity Express 
 end mailbox

voicemail callerid  ! Optional - Used to enable playing of caller ID information from incoming voice-mail messages

!Basic

voice register global
 mode cme
 source-address 192.168.1.32 port 5060
 max-dn 10
 max-pool 2
 authenticate register

!Advanced
! Other Features
 timeouts interdigit 5
 tftp-path flash:
 ntp-server 192.168.1.250 mode directedbroadcast
 timezone 1 ! type ? for a list
 date-format D/M/Y
 camera ! to enable camera
 video ! to enable video
! secondary dial-tone on SIP is not supported

voice class codec 1 ! Just to prioritize the codec
 codec preference 1 g711ulaw
 codec preference 2 ilbc
  video codec h264


voice service voip
 no ip address trusted authenticate
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 sip
  bind control source-interface Vlan2 ! voice vlan or voice sub-interface 
  bind media source-interface Vlan2 ! voice vlan or voice sub-interface
  registrar server

voice register dn  1
 number 2001
 name HQ Phone 1
!
!
voice register pool  1
 id mac <MAC>
 type 9971
 number 1 dn 1 ! Button 1, DN 1 (2001) configured above
 description +14165552001
 username hqone password cisco
 no call-waiting
 dtmf-relay sip-kpml rtp-nte
 voice-class codec 1 ! g711 would be preferred based on the 'voice class codec' configured
 busy-trigger-per-button 1 ! second call would be busy
 no vad
 camera
 video
!

1. Infrastructure
Make sure basic infrastructure is there:
For example: VLAN, DHCP, NTP, etc

2. Basic CME Config

telephony-service
 max-ephones 10
 max-dn 30
 ip source-address 10.1.1.1 ! voice vlan/loopback address
 create cnf-files

Power on phones in sequence.
To verify,
 sh ephone

3. Configuring Phone Numbers

ephone-dn 1
 number 1001

ephone-dn 2
 number 1002

ephone 1
 mac-address <MAC>
 button 1:1 ! Button 1, DN 1 (configured above)
 restart

ephone 2
 mac-address <MAC>
 button 1:2 ! ! Button 1, DN 2 (configured above)
 restart

4. Changing Display Message on Phone

telephony-service
 system message Jalson International Inc

5. Dual Channel

ephone-dn 3 dual-line
 number 1003

ephone 1
 button 1:1 2:3
 restart

To verify: sh ephone

6. Seconadry Dial Tone

telephony-service
 secondary-dialtone 9

7. Call Blocking

telephony-service
 after-hours block pattern 1 91900 7-24
 after-hours block pattern 2 9011
 after-hours ???

ephone 2
 after-hour exepmt

Ephone 2 will be able to dial 91900 (Premium) and 9011 (International)

8. Call Distribution -- shared lines

ephone-dn 7
 number 1077

ephone 1
 button 2:7

ephone 2
 button 2:7

9. Timezone

telephony-service
 time-zone 43 ! type ? aftertimezone to see the list.

10. Date Format

telephony-service
  date-format dd-mm-yy

11. Call Transfer/Call Forward Allow to Any Pattern

telephony-service
 call-forward pattern .T ! to forward to any pattern
 transfer-pattern .T ! to transfer to any pattern including PSTN
 transfer-system full consult ! full-consult - won't allow blind transfer