CIPT1

1LAB 1: Call Manager Installation

1Experiment (i): Installing a Publisher

2Experiment (ii): Installing a Subscriber

2Experiment (iii) Enabling various services

4LAB 2: Initializing IP Phones

4Experiment (i): Configuring a Switch For VoIP support Voice VLAN & PoE configuration

6Experiment (ii): Initializing IP Phone

7Experiment (iii): Adding an IP Phone using BAT

9Experiment (iv): Hardening IP phones

10LAB 3: User Management

10Experiment (i): Managing User Accounts

11Experiment (ii): Managing User accounts using BAT

12Experiment (iii): Configuring Third party SIP Phones

14Experiment (iv): Configuring SIP Digest Credentials

15Experiment (v) LDAP Syncronization

16Experiment (vi) LDAP Authentication

17LAB 4: Gateways

17Experiment (i) MGCP Gateways

20Lab 5: Call routing

20Experiment (i) Configuring Route Patterns

21Experiment (ii) Configuring Backup Route

23Experiment (iii) Digit manipulation

24Experiment(iv) Hunt Group

26Experiment (v) Partition and CSS

30LAB 6: Media Resources

30Experiment (i): Configuring Software Conference Bridge

31Experiment (ii): Configuring Meet Me Conference Bridge

31Experiment (iii): Music On Hold

32Experiment (iv): Multicast MOH

33Experiment (v): Hardware Conference Bridge

35LAB 7:User Features

35Experiment (i): Basic features

38Experiment (ii): Configuring Presence & Speed Dials

41Lab 8: Voicemail Integration

LAB 1: Call Manager InstallationExperiment (i): Installing a Publisher

Aim: To install cisco unified communication manager.Steps to configure.

Insert the disk and start the installation.

Step 1: Select the CUCM as the product to be installed in Product Deployment Selection window. Click yes in next window to proceed with installation.

Step 2: Select Proceed in the Platform Installation Wizard window.

Step 3: Select No in Apply patch window since we are going to install new cucm. Select No in Import Window Data section & press continue.

Step 4: Set appropriate timezone in Timezone configuration window.

Step 5: Select Yes for automatic speed negotiations with NIC card, No for DHCP configuration. Enter the name(CUCM Publisher), ip address(10.0.0.10),subnet mask(255.0.0.0) and default gateway(10.0.0.1) in Static network configuration window.Step 6: Enter the OS Administrator ID(osadmin) and password(p@ssw0rd) in Administration Login Information. Enter organization details(name, unit, location, state, country) in the Certification Information Window.

Step 7: Select Yes in first node configuration window, select No in Network time Protocol client Configuration Window and set the time & date manually in next window. Set the password in Database Access Security Configuration.

Step 8: Select no to SMTP configuration. Set the username and password in Application user configuration window for GUI access of CUCM and click OK to finalize the platform Configuration. Installation starts.

Verification:1) Access the CUCM through web browser using the IP http://10.0.0.10. And login into the CUCM administration window.

2) Login into OS Administration window using osadmin username and password.

Experiment (ii): Installing a SubscriberAim: To install a Subscriber in a CUCM cluster.

Steps to Configure:

Step 1: A) Go to System > Server to add the new subscriber on the publisher. Click the Add New

Button and enter the subscribers ip address and Save.B) similarly change the name of publisher to its ip address for disabling DNS dependency.

Step 2: Insert the CD and start the Subscriber installation process. First 6 steps are same as previous experiment. In static network configuration Window enter the following information.Name: CUCM SubscriberIp address: 10.0.0.20Subnet Mask: 255.0.0.0Gateway IP: 10.0.0.1Step 3: Select No in the first node configuration Window since we are going to install subscriber and Ok in warning window.

Step 4: Enter Publishers information in first node configuration window and the installation will proceed.

Verification:1) Access the subscriber using the ip 10.0.0.20, login using the publishers user name and password.

Experiment (iii) Enabling various services

Aim: To enable various basic services in CUCM such as DHCP server configuration.

Steps to configure.

Step 1: Log into the CUCM Serviceability.

Step 2: In the CUCM Serviceability, go to Tools > Service Activation. From the list of services of the server, select 10.0.0.10, check the Cisco CallManager, the Cisco Tftp, and the Cisco DHCP Monitor Service to activate them.Step 3: Configure DHCP Server.

In the CUCM Administration navigate to System > DHCP > DHCP Server and click the Add

New button. Select 10.0.0.10 from the Host Server drop-down list, set the Primary TFTP Server IP Address (Option 150) to 10.0.0.10 and Save.

Step 4: Configure DHCP Subnet.Go to System > DHCP > DHCP Subnet and click the Add New button. Select the newly created DHCP Server 10.0.0.10 and enter the following parameters:

Subnet IP Address: 10.0.0.0

Primary Start IP Address: 10.0.0.100

Primary End IP Address: 10.0.0.200

Primary Router IP Address: 10.0.0.1

Subnet Mask: 255.0.0.0Verification:

Go to control Tools > control center-Feature Services and check the activation status tab.LAB 2: Initializing IP Phones

Experiment (i): Configuring a Switch For VoIP support Voice VLAN & PoE configurationAim: To configure a Voice Vlan & enable Power Over Ethernet on a CISCO L2 switch.Steps to configure:Step 1: Create VLANs for Voice & Data. The procedure to create Voice /Access Vlan is the same. Only when we associate it to an interface, we specify it to be a Voice or Access VLAN.

Switch(config)# vlan

Switch(config-vlan)#name

Switch(config)#interface

Eg:

Switch(config)# vlan 2

Switch(config-vlan)#name voice

Switch(config-vlan)#exit

Switch(config)# vlan 3

Switch(config-vlan)#name data

Step 2: Associate the VLANs to an interface

Switch(config-if)#switchport voice vlan

Switch(config-if)#switchport access vlan

Lets choose an interface where we have connected an IP phone & a computer (to the IP Phone). We shall now configure these two devices to belong to different vlans.

Switch(config-if)#switchport voice vlan 2

Switch(config-if)#switchport access vlan 3Verification :

Lets now check the VLAN table. Observe that Fa0/4 is displayed against 2 VLANs Voice Vlan 2 & Access Vlan 3.switch#show vlan

VLAN Name Status Ports

---- -------------------------------- --------- -------------------------------

1 default active Fa0/1, Fa0/2, Fa0/3, Fa0/5,Fa0/6, Fa0/7,Fa0/8,

Fa0/9,Fa0/10,Fa0/11,Fa0/12,Fa0/13,Fa0/14, Fa0/15, Fa0/16, Fa0/17, Fa0/18, Fa0/19 Fa0/20, Fa0/21, Fa0/22, Fa0/23, Fa0/24, Gi0/1,Gi0/2

2

voice active Fa0/4

3

data active Fa0/4

In the second part of our experiment, lets look at configuring inline power options. Switch(config)#interface

Switch(config-if)#power inline ?

auto

Automatically detect and power inline devices

consumption Configure the inline device consumption

never Never apply inline power

static

High priority inline power interface

The above options show enabling / disabling PoE & how to automatically detect phone & supply requisite power or statically define power (in milliwatts) - as shown below.

Eg:

Switch(config-if)#power inline consumption 6000

%CAUTION: Interface Fa0/4: Misconfiguring the 'power inline

consumption/allocation' command may cause damage to the switch and void

your warranty. Take precaution not to oversubscribe the power supply.

It is recommended to enable power policing if the switch supports it.

Refer to documentation.

Verification :

Switch#show power inline

Available:370.0(w) Used:6.3(w) Remaining:363.7(w)

Interface Admin Oper Power Device Class Max

(Watts)

--------- ------ ---------- ------- ------------------- ---------------- --------------

Fa0/1 auto off 0.0 n/a n/a 15.4



Fa0/4 auto on 6.3 IP Phone 7940 n/a 15.4





Interface fa0/4 is connected to 7940 phone & is supplied 6.3 W by the PoE switch.Experiment (ii): Initializing IP PhoneAim: To initialize an IP phone using CUCM - Automatically and Manually.Configuration StepsThe pre-requisite for Auto/Manual Registration of an IP Phone is to have the Device Pool configured. In our experiment, well use the Default Device Pool.

Step 1: Verify the Default Device pool.

Go to System > Device Pool and click Find, Select the Default device pool and verify the following

Cisco Unified Communications Manager Group: Default Date/Time Group: CMLocal (use the date, time and timezone of the CUCM) Region: Default SRST Reference: Disable

Step 2: Lets now Activate & add IP Phone through Auto Registration (for 10 IP phones with extensions 1001 to 1010).Go to System > Cisco Unified CM. Click the Find button and select Publisher, in Auto-registration Information pane set 1001 for the starting directory number and 1010 for the ending directory number. Verify that the Auto-Registration Disabled check box is unchecked and click Save.

Step 3: Add IP Phone Manually.

Go to Device > Phone Select Add New select the phone type as 7940 and Protocol as SCCP. And enter the following Details,

Mac Address: 0023.EB54.BDF9 Description: 7940 phone Device Pool: Default Phone Button Template: Standard 7940 SCCP Device Security Profile: Cisco 7940 Standard Non-Secure ProfileSelect Line [1]-Add a New DN and set the Directory Number as 1011 and click Save.

Note:

1. Mac address of the phone will be printed at the back of the phone. The other way is to navigate Settings > Model Information of IP Phone.2. Another interesting method for Large scale Deployment is to enable Auto registration for all phones initially & subsequently change the DNs manually as required.Verification:

1) Auto Registration.

Connect an ip Phone to the CUCM & it will automatically register itself with the CUCM with a number say 1001(numbers will be randomly assigned to phones).

2) Manual configuration.a) Now connect the IP phone that we manually configured with the CUCM. The phone will register with the number 1011.Place a call to 1001 & the call should go through. b) Now connect a pc to the ip phone and capture the packet using wire shark. Filter the RTP packets and save it as .au file. Now u can playback the conversation.

Experiment (iii): Adding an IP Phone using BAT

Aim: To add many IP phones in a single step using Bulk Administration Tool.Steps to configure: Step 1: Enable BAT Services.

Go to Tools > Service Activation. Database and Admin Services area, activate the Cisco Bulk Provisioning Service.Step 2: Configure IP Phone Template.

Go to Bulk Administration > Phones > Phone Template and click Add New select the phone type as 7940 and configure as follows,

Template Name: 7961BAT Description: 7961 template Device Pool: Default

Phone Button Template: Standard 7961 SCCP

Device Security Profile: Cisco 7961 Standard SCCP Non-Secure Profile

Select the Line [1] link in the left column, then enter line1 for Line Template Name.

Step 3: Create CSV file.

Go to Bulk Administration > Upload/Download Files and then click Find. Check the bat.xlt file and click Download. In the Phones spreadsheet tab, click the Phones radio button and then the Create File Format button. In the new dialog window, add Directory Number in the Line Fields pane, and then Click Create button and then Yes on the pop-up window to overwrite the existing file. On the Phones page, enter five new IP phones with the parameters that followDescription Directory Number mac addressIPPhone1

2006

3006.111a.1243IPPhone2

2007

3007.222b.1235IPPhone2

2008

3008.2354.ab12IPPhone4

2009

3009.abc1.de34IPPhone5

2010

3010.edac.9012click the Export to BAT Format button. Save the new file as testbat.

Step 4: Validate the IP Phone Template and CSV File

Go to Bulk Administration > Upload/Download Files and click Add New and add the created testbat file. Choose Insert Phones Specific Details from the Select Transaction Type drop-down menu Go to Bulk Administration > Phones > Validate phones. Select the previously uploaded CSV file. From the Phone Template Name menu, choose the BAT template and click Submit. Go to Bulk Administration > Job Scheduler and click Find. In the job list shown, click the Job Id link that has Validate Specific Phones in the Description column. The job results should display the validation status without errors. Go to Bulk Administration > Phones > Insert phones and select File that we have uploaded testbat, Phone Template Name 7961BAT and check Run immediately Check box.Verification: Now connect the phone with Mac 3006.111a.1243. It should register with call manager and get the number 2006.

Experiment (iv): Hardening IP phones

Aim: To harden ip phone for enhanced security.

Steps to configure.

Go to Device > phone and select phone 1011 and do the following, in product specific configuration,

Disable Gratuitous ARP Disable PC Voice VLAN Access

Disable PC Port Disable Settings Access Disable Web AccessVerification:

1) Now try to access the phone using its ip through the browser. The phone page should not be displayed.

2) Try to navigate through the settings button - it will not show up anything as before.

3) Now connect a pc to the IP phone - the system will not come up.

LAB 3: User Management

Experiment (i): Managing User Accounts

Aim: To create and assign various roles to End Users so that they may be able to manage user facing features on phones. In this experiment, well configure 2 users one to only view details of configured features & the other with capability to modify. Steps to configure.

Step 1: Create End users : Lets create user1 & user2Go to User Management > End User and click Add New. Configure a user with the attributes that follow, and save the account.

User ID: User1Password: p@ssw0rdPIN: 54321Last name: Phone1First name: User1Create user2 similar to the above.

Step 2: Creating user Groups & Roles. Go to User Management > User Group click add new, create 2 groups UserReadOnly, UserReadWrite.

Go to User Management > Roles and copy Standard CCM End User Save it as Read only and remove the write privileges. Click Roles button and new roles to groups as follows,

UserReadOnly: Standard End Users , ReadOnly UserReadWrite: Standard CCM End User, Standard End Users.Step 3: Add devices and Groups to Users

Go to User Management > Enduser select find and select user1, associate Device 1001 in Device Association Pane, similarly associate phone 1011 to user 2. Add groups by clicking Add user Group button in Permission Information Pane, add UserReadOnly, UserReadWrite to user1 & user2 respectively.Verification:Go to the user administration page using the format http://ipaddress/ccmuser

Login as user 1 and view all the details & observe you cant change any features.

Login as user 2 and configure call forward and any other user facing features. Youll be allowed to do so.

Experiment (ii): Managing User accounts using BAT

Aim: To add users using BAT.Steps:Step 1: Enable Cisco Bulk Provisioning Service.

Go to Tools > Service Activation.Database and Admin Services area, select the Cisco Bulk Provisioning Service.

Step 2: Configure User Template.

Go to Bulk Administration > Users > User Template and click Add New button & configure as follows,User Template Name: RJPUser Locale: English,US

User Group: Standard User GroupStep 3: Create CSV file.

Go to Bulk Administration > Upload/Download Files and the click Find. Check the check box next to the bat.xlt file and click the Download Selected button.

In the user tab add the following

UserId

FirstnameLastnamePassword Pin

User3

user

3

p@ssw0rd12345

User4

user

4

p@ssw0rd54321

click the Export to BAT Format button. Save the new file as userbat.

Step 4: Insert Users

Go to Bulk Administration > Upload/Download Files and click the Add New button. In the File text box, click the Browse button and locate the file userbat.txt. Choose from Select the Target drop-down list box Users and from the Select Transaction, in Type drop-down list box Insert Users.

Go to Bulk Administration > Users > Insert Users. Choose in the File Name field userbat. Select the user template that you created (RJP) from the User Template Name dropdown list box. Check the Run Immediately radio button and click Submit.

Verification:

Go to User administration page using the format http://ipaddress/ccmuser,login as user3. Youd be allowed to login.Experiment (iii): Configuring Third party SIP PhonesAim: To configure a Third Party SIP Phone Xlite in this case. We associate a SIP Phone with a CUCM end user as MAC address is not considered.Steps to configure.

Step1: Configure End User.Go to User Management > End User and click Add New. Create a new account with following parameters User ID: SIPPhone Last Name: User Password: p@ssw0rd PIN: 12345Step 2: Create SIP Phone.

1) Choose Device > Phone and click Add New. For the Phone Type, choose Third-party SIP Device Basic Configure these parameters: MAC address: 123212311231(any dummy entry would do) Description: SIP-Phone Device Pool: Default Phone Button Template: Third-party SIP Device (Basic) SIP Phone Security Profile: Third-party SIP Device Basic Standard SIP Non-Secure Profile SIP Profile: Standard SIP Profile Digest User: SIPPhone2) Select Line[1]-Add a new DN and set the directory number as 1004.

Step 3: Configure SIP phone

Launch the X-Lite SIP phone. Right click on the phone and select SIP Account Settings and configure following details.

Display name: SIP User.

User name: 1004 Authorized User name: SIPPhone Domain:10.0.0.10Verification:

We can visually observe that the SIP phone is now added to the CUCM as shown above. Place a call to 1011 from 1004 & the call will go through.Experiment (iv): Configuring SIP Digest CredentialsAim: To enable security feature by enabling authentication using SIP Digest Credentials. Steps to Configure.

Step 1: Configure Phone Security ProfileGo to System > Security Profile > Phone Security Profile and click Find and Copy Third-party SIP Device Basic Standard SIP Non-Secure Profile. Change the profile name and description to Digest Authentication. Activate the Enable Digest Authentication check box.

Step 2:Configure End User.

Go to User Management > End User and click Add New. Create a new account with following parameters

User ID: DigestPhone Last Name: User Password: p@ss PIN: 12345 Digest Credentials: p@ssw0rdStep 2: Create SIP Phone.

3) Choose Device > Phone and click Add New.For the Phone Type, choose Third-party SIP Device Basic Configure these parameters: MAC address: 123212311231(dummy) Description: SIP-Phone

Device Pool: Default

Phone Button Template: Third-party SIP Device (Basic) SIP Phone Security Profile: Digest Authentication SIP Profile: Standard SIP Profile

Digest User: SIPPhone4) Select Line[1]-Add a new DN and set the directory number as 1005.

Step 3: Configure SIP phone

Right click on the phone and select SIP Account Settings and configure following details.

Display name: SIP User.

Password : p@ssw0rd User name: 1005 Authorized User name: SIPPhone Domain:10.0.0.10Verification:

Initialize the SIP phone without entering password the phone will not register. Now Enter the password & the phone will register.

Experiment (v) LDAP SyncronizationAim: To synchronize and manage CUCM users from LDAP directory database.

Steps to configure:

Step 1: Configure ADS on server 2003 create an OU with 2 users & assign it an Admin. Login in to windows 2003 server go to Start > Programs > Administrative Tools > Active Directory Users and Computers.

Right click DC and create new User Admin with password p@ssw0rd,Right click on the user and make it a member of administrator group.

Add a New Organizational unit with 2 users and delegate the control of this OU to Admin user created above. Let the OU be named test.Step 2: Configure LDAP System on CUCM & Activate DirSync Services. Go to Tools > Service Activation in CUCM serviceability and activate Cisco Dirsync Service. Go to System > LDAP > LDAP System and check enable synchronization check box , Select Microsoft active directory as LDAP Server type and LDAP attribute for UserID as sAMAccountName.

Step 3: Configuring LDAP Directory Go to System > LDAP > LDAP Directory and configure the following

LDAP Configuration Name: LDAPSYNCLDAP Manager Distinguished Name: AdminLDAP password: p@ssw0rdLDAP User Search Base: ou=test,dc=rjp,dc=edu

LDAP Server IP: , perform the full sync once.

Verification:

Now go to User Management > End User and click find, notice that the users from DC will have been updated and old users configured in database will be shown as inactive (if not present in DC)

Add New button & delete button would have vanished, meaning no users can be created/deleted from CUCM. Select any of the users and Change the password. Itll allow to do so because only user account has been enabled for LDAP Sync & not Authentication.Experiment (vi) LDAP AuthenticationAim: To Authenticate users using LDAP & not CUCMSteps to Configure:

Step 1: Configure LDAP System as before.Go to System > LDAP > LDAP System and check enable synchronization check box , Select Microsoft active directory as LDAP Server type and select LDAP attribute for UserID as sAMAccountName.

Step 2: Configuring LDAP Directory Go to System > LDAP > LDAP > LDAP Directory and configure the following

LDAP Configuration Name: LDAPSYNCLDAP Manager Distinguished Name: admin( he must be delegated the control to access the ou)

LDAP password: p@ssw0rdLDAP User Search Base: ou=test,dc=rjp,dc=edu


Step 4: Configuring LDAP Authentication.

Go to System > LDAP >LDAP Authentication,Enable the LDAP Authentication for End Users check box

LDAP Manager Distinguished Name: [email protected] password: p@ssw0rdLDAP User Search Base: ou=test,dc=rjp,dc=edu


Verification: Now go to CUCM User Management > End User and select any one of the user and try to change the password. Itll not allow password changes. Terminate the connection between windows 2003 and CM and try to log into the user webpage, the authentication will fail.LAB 4: Gateways

Experiment (i) MGCP Gateways

Aim: To register an MGCP gateway, configure its analog ports and E1 Trunks from CUCM.Note: In this lab, we choose 2811 Voice Gateway & some of its ports for configuration. Please ensure you make the correct selection based on the Voicde Gateway you are working with.Steps to configure : The first 3 steps are configurations on the CUCMStep1: Configure MGCP gateway: First create the Gateway configuration on the CUCMGo to Device > Gateway click Add new select the 2811 router and select MGCP protocol, configure the Domain Name as R1,device pool Default, Module in Slot 0 as NM-4VWIC-MBRD and click save. Configure VWIC3-4FXS/DID, VWIC-2MFT-E1 in subunit 0 & 1 respectively.Step 2: Configure FXS Ports.In this step we will be configuring and assigning numbers to FXS Ports to which analog phones will be connected.Go to Device > gateway click find and select router R1,click subunit 0/0/0 and configure device pool as Default and save. Select line 1 and add new directory number as 1001 and save. Similarly configure 1002 for the FXS port.

Step 3:Configure E1 trunk.In this step we will be configuring the ISDN PRI trunk for the above Gateway.

Go to Device > gateway and select router R1, click subunit 0/1/0, configure device pool as default and configure channel selection order as either top down or bottom up.

Step4:CLI Configuration on the Gateway.In this step we will be handing over the control of Gateway to CUCM so that it will download the configuration file from CUCM.Configure the hostname of router as R1,enter following commands

Mgcp

ccm-manager config server

- to specify tftp server IPccm-manager config

- to pull config from CUCMVerification 1:

The below shown output is seen when router downloads the configuration files from CUCM.

Loading R1.cnf.xml from 20.0.0.10 (via FastEthernet0/0): !

[OK - 7395 bytes]Verification 2: Place a call from 1001 to 1002 & the call should go through.

Go to Device > gateway and select router R1, click subunit 0/0/0 - it should be registered with CUCM.

Verification 3: On GatewayR2#sh isdn status

Global ISDN Switchtype = primary-4ess

%Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 0. Layer 3 output may not apply

ISDN Serial0/1/0:15 interface

dsl 0, interface ISDN Switchtype = primary-net5

L2 Protocol = Q.921 0x0000 L3 Protocol(s) = CCM MANAGER 0x0003

Layer 1 Status:

ACTIVE

Layer 2 Status:

TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED

Layer 3 Status:

0 Active Layer 3 Call(s)

Active dsl 0 CCBs = 0

The Free Channel Mask: 0x80000FFF

Number of L2 Discards = 0, L2 Session ID = 1

Total Allocated ISDN CCBs = 0

Above output displays layer 3 is binded to cucm.

Verification 4: On the GatewayR1# Show Mgcp

MGCP Admin State ACTIVE, Oper State ACTIVE - Cause Code NONE

MGCP call-agent: 20.0.0.10 2427 Initial protocol service is MGCP 0.1

MGCP validate call-agent source-ipaddr DISABLED

MGCP validate domain name DISABLED

MGCP block-newcalls DISABLED

MGCP send SGCP RSIP: forced/restart/graceful/disconnected DISABLED

MGCP quarantine mode discard/step

MGCP quarantine of persistent events is ENABLED

MGCP dtmf-relay for VoIP is SDP controlled

MGCP dtmf-relay for voAAL2 is SDP controlled

MGCP voip modem passthrough mode: NSE, codec: g711ulaw, redundancy: DISABLED,

MGCP voaal2 modem passthrough disabled

MGCP voip modem relay: Disabled

MGCP T.38 Named Signalling Event (NSE) response timer: 200

MGCP Network (IP/AAL2) Continuity Test timer: 200

MGCP 'RTP stream loss' timer disabled

MGCP request timeout 500

MGCP maximum exponential request timeout 4000

MGCP rtp unreachable timeout 1000 action notify

MGCP gateway port: 2427, MGCP maximum waiting delay 3000In above output the admin and operational states are Active.Lab 5: Call routing

Experiment (i) Configuring Route Patterns

Aim: To route PSTN Calls using route patterns in CUCM.

Steps to configure:

Step1: Configure E1 trunk

In this step we will configure the MGCP gateway and E1 trunk.

Go to Device > Gateway Add new > 2811 router and select MGCP protocol, configure the Domain Name as R1,device pool Default, Module in Slot 0 as NM-4VWIC-MBRD and click save. Configure VWIC3-4FXS/DID, VWIC-2MFT-E1 in subunit 0 & 1 respectively.

Go to Device > gateway select router R1, click subunit 0/1/0, configure device pool as default and configure channel selection order as either top down or bottom up.

Step 2: configuring Route pattern

In this step, well create a Route Pattern in CUCM. Go to Call Routing > Route/Hunt > Route Pattern, Add new and enter following details,

Route Pattern: 9.@Numbering Plan: NANPGateway/Route List: S0/SU1/DS1-0@R1(e1 trunk)Route Option: Route this patternCall Classification: OffNet Discard Digits: PreDot, and save the configuration

Verification:

Now place the call to 92431005432 from 1001, where 9 is the access code. 9 will be stripped off automatically before the call is forwarded to the PSTN (as per the above configuration).

Experiment (ii) Configuring Backup Route

Aim: To implement backup routes (redundancy) using Route List & Route Group in CUCM.Steps to Configure

Step1: Configure MGCP Gateway and Trunk.

Go to Device > Gateway click Add new select the 2811 router and select MGCP protocol, configure the Domain Name as R1,device pool Default, Module in Slot 0 as NM-4VWIC-MBRD and click save. Configure VWIC3-4FXS/DID, VWIC-2MFT-E1 in subunit 0 & 1 respectively.

Go to Device > gateway click find and select router R1,click subunit 0/1/0 ,configure device pool as default and configure channel selection order as either top down or bottom up. Similarly configure for router R2.

Step 2: Configure Route Group and route list.

Here we will be configuring a Route Group with the primary and backup Link to the PSTN and will be adding it to a Route List.

Go to Call Routing > Route/Hunt > Route Group, add new route group with the following details,

Route Group Name: BackupRgDistribution Algorithm: TopDown

Available Devices: S0/SU1/DS1-0@R1, S0/SU1/DS1-0@R2 (Primary link via R1 should be the first available device in route group members) save the configuration.

Go to Call Routing > Route/Hunt > Route List, add new list with name RL, Cisco Unified Communications Manager Group as Default and save it, Click the Add Route Group Button and select the BackupRg group and save the configuration.Step 3: Configure Route Pattern.Here we add a Route Pattern to access PSTN phones. The procedure is the same as before. Only that we select the RouteList instead of the Gateway.Go to Call Routing > Route/Hunt > Route Pattern ,click Add new and enter following details,

Route Pattern: 9.@Numbering Plan: NANP

Gateway/Route List: RLRoute Option: Route this pattern

Call Classification: OffNet

Discard Digits: PreDot, and save the configuration.

Verification:

Now place a call to 92431005432 & the call should go via primary link. Use show isdn status command to verify the same.

Now disconnect the primary link and place the call. It should go via backup router R2.Experiment (iii) Digit manipulation

Aim: To implement Digit manipulation techniques using external phone number mask, translation pattern & various transformation techniques.

Task1:

In this task we are going to translate our extension into e.164 standard when placing a call to the PSTN. This can be done in 3 ways which is described as follows. (After configuring each method place a call to verify)

External Phone Number Mask : Go to Device > Phone click find and select 1001, go to the line setting of the phone and apply external phone number mask as 55523451001. Go to Call Routing > Route/Hunt > Route Pattern and check the use Calling Party External Phone Number Mask check box.

Route Group - Called party transformation mask : Go to Call Routing > Route/Hunt > Route List and select the Route List RL and select the Route Group BackupRg from Route List Details pane and apply called party transformation mask as 5552345XXXX else apply Prefix DN as 5552345. Route Pattern - Called party transformation mask : Go to Call Routing > Route/Hunt > Route Pattern and select the pattern 9.@ and configure called party transformation mask as 5552345XXXX else apply Prefix DN as 5552345.Task 2:

When a call is received via PSTN, called number would be a 10 digit number with area code and country code. This should be stripped off before forwarding it to a 4 digit internal extension. Access code 9 should also be added to the calling number so that users can redial without dialing the access code. This is done with the Translation Patterns Calling/Called Party Transformation Mask as shown below.

Go to Call Routing > Translation pattern and click Add New button, create new Translation pattern:555444XXXXCalling Party Transformation Prefix DN: 9Called Party Transformation Mask: XXXX Verification:

For each experiment, place a call and verify the Calling and Called party numbers. Experiment(iv) Hunt Group

Aim: To configure a single board number of a call center & route incoming calls to one of many extensions using Hunt Pilot.Experiment : Assume a call center has 6 lines but only one Board Number (say 1000). Whenever a call is received on the board number, it has to be forwarded to one of available 6 extensions (1001 to 1006) according to preconfigured algorithms. In the below case, we organize the extensions in 2 groups (Hunt Groups) & map them to the Hunt Pilot (the Board Number) using Hunt List. (Please try out the other available algorithms apart from the one discussed below).

Step 1: Configure Line groups LG1 & LG2.

Go to Call Routing > Route/Hunt > Line Group and create a new line group with following details,

Line Group Name: LG1RNA Reversion Tine :5Distribution Algorithm: Top DownNo Answer: Try next member; then, try next group in hunt listBusy: Try next member, but do not go to next goupNot Available: Try next member; then, try next group in hunt list, select 1001,1002,1003 from available DN and click Add to Line Group.

Similarly configure line group LG2 as follows

Line Group Name: LG2RNA Reversion Tine :5Distribution Algorithm: BroadcastNo Answer: Try next member; then, try next group in hunt listBusy: Try next member, but do not go to next groupNot Available: Try next member; then, try next group in hunt list, select 1005,1006 from available DN and click Add to Line Group.

Step 2: Configure Hunt listGo to Call Routing > Route/Hunt > Hunt List and add new hunt list with name HL1,select CUCM Group as Default, check the Enable this Hunt List option and add line group LG1, LG2 by clicking Add Line Group button.

Step 3: Configure Hunt PilotGo to Call Routing > Route/Hunt > Hunt Pilot and add new pilot as follows

Hunt pilot: 1000Hunt List: HL1Forward Hunt No Answer: 1004Forward Hunt Busy: 1004, and Save.

Verification:

Place a call to 1000 and dont answer the call from any of the internal extensions. The call should get forwarded through 1001,1002,1003 respectively and after ending LG1 both 1005, 1006 should ring simultaneously (broadcast algorithm used in LG2).

Now keep all the Hunting extensions busy by keeping them in off hook state and from pstn place a call to 1000. The call should land on 1004 whichs the final forwarding number (Forward Hunt No Answer).Experiment (v) Partition and CSS

Lobby-1001 & 1002Receptionist- 1003Manager- 1004MD- 1005

Aim: To implement Class of Service (Class of Restriction) using Partition & CSS for restricting access. This is used to control the telephony charges. This feature is very similar to Access lists which are used to define restrictions. Partitions (similar to locks) are first created & then CSS (Calling Search Space) are created with select Partitions as members (keys). CSS is configured on the Device/Translation Pattern that seeks to access a phone/trunk & Partitions are configured on the seeked component like Route Pattern/Phone.

The Requirement:

i) Lobby phones (1001&1002) are allowed to call other Lobby phones & Receptionist phone.

ii) Reception phone (1003) can only call Lobby & Manager phones.

iii) Manager phone (1004) can call MD, Lobby & Receptionist phones. He can access PSTN phone only during office Hours (10am to 6pm). After working hours FAC should be implemented.iv) MD (1005) can access Manager, MD, Lobby, Receptionist & PSTN phones.

Steps to Configure:Step 1: Create Partitions.Go to Call Routing > Class of Control > Partition and create new partitions as follows,

1) MD

2) Manager

3) PSTN

4) PSTN_FAC

5) PSTN_TOD

6) Receptionist

7) Lobby

Step 2: Create CSS & include Partitions as per the below tableGo to Call Routing > Class of Control > CSS and create CSS as follows,

MDManagerReceptionistLobbyPSTNPSTN_FACPSTN_TOD

Manager_CSSXXXXXX

Lobby_CSSXX

MD_CSSXXXXX

Rec_CSSXXX

Incoming_CSSXXXX

Step 3: Configuring Time of the DayTime period is defined for office hours so that Managers PSTN access is allowed by default during this time. This is done by associating the time period with the PSTN_TOD partition. Go to Call Routing > Class of Control > Time Period add new period as follows

Name: WorkHours

Time of Day Start: 10:00

Time of Day End: 18:00

Go to Call Routing > Class of Control > Time Schedule and add new schedule as Worktime and move WorkHours from available time period to selected time period.

Go to Go to Call Routing > Class of Control > Partition PSTN_TOD and apply Newly created time schedule Worktime to it.Step 4: Apply Partition to PhonesGo to Device > Phone, find 1001 and apply Lobby partition to the line 1 of 1001. Similarly apply appropriate partitions to all other phones.

Step 5: Applying CSS to Phones.Go to Device > Phone, 1001 and apply Lobby_CSS either at the device or at the Line 1 of the phone. Similarly apply the appropriate CSS to all other phones.

Step 6: Applying Partition to Route Patterns.Go to Call Routing > Route/Hunt >Route Pattern, Add 3 9.@ Route Patterns with different Partitions. One without time association, one with time association & one with FAC enabled.

Go to Call Routing > Route/Hunt > Route Pattern - Add new and enter following details,

Route Pattern: 9.@Partition: PSTNNumbering Plan: NANP



Discard Digits: PreDot,

Route Pattern: 9.@Partition: PSTN_FACNumbering Plan: NANP



Discard Digits: PreDot,

Route Pattern: 9.@Partition: PSTN_TODNumbering Plan: NANP



Discard Digits: PreDot.Step 7: Configuring Forced Authorization Code. Go to Call Routing > Forced Authorization Code, add new as follows

Authorization Code Name: FACAuthorization Code: 12345

Authorization Level: 4 Go to Call Routing > Route/Hunt >Route Pattern, and find route pattern 9.@ with partition PSTN_FAC, and check the Require Forced Authorization Code checkbox and set minimum level to 4.Note:

Make sure that in Manager_CSS, PSTN_TOD partition is listed first & PSTN_FAC partition next.

Step 7: Configuring CSS for Incoming callsGo to Device > Gateway, find gateway R1 and apply the Incoming_CSS in the E1 trunk.

Note : If a translation pattern is configured for incoming calls, a separate CSS will be required to be applied on it.

Verification:i) Place calls from all the phones and check that the above conditions are met.

ii) Place a PSTN call from MD phone & it goes through at all times.

iii) Place a PSTN call from Manager phone during office hours & the call should go through. Change the time period to 0:00 to 0:15 (non-office hours) and place a call to PSTN. It will match the route pattern with a partition of PSTN_TOD & the call would not be permitted. Next itll try the Route Pattern with Partition PSTN_FAC & this will ask for FAC code. Enter the code & the call would go through.LAB 6: Media ResourcesWe shall learn how to configure the various Media Resources available in the CUCM. We shall configure Software / Hardware Conference Bridges & Music On Hold (MoH) in the below section.

Experiment (i): Configuring Software Conference Bridge

Aim: To configure Software conference bridge (supports only g711 conference) Before configuring try to place a conference call from any phone & it displays No Conference Bridge.

Go to Media Resource > Conference Bridge & see that the registration state of CFB_2 (Software Conference Bridge) is unknown.Step1: Enabling Software Conference Bridge.Go to CUCM Serviceability > Tools > Service Activation and activate Cisco IP Voice Media Streaming App.

Verification: Now establish an ad-hoc Conference (using the conference softkey) & the conference call will succeed.

Go to Media Resource > Conference Bridge & observe that the CFB_2 is in Registered State.Experiment (ii): Configuring Meet Me Conference BridgeAim: This involves configuring a Number as a Meet me conference number & accessing this number to setup a conference. Steps to configure:

Go to Call Routing > Meet-Me Number/Pattern and add new meet me number 1122.Verification:From 1004 use Meet Me Softkey (displayed after using the more softkey) and dial 1122. Now, Dial 1122 from extensions 1005 and 1006 to join the MeetMe conference.

Experiment (iii): Music On HoldAim:

To upload an MoH file and configure it for use by some phones as their Music on Hold tunes.

Step 1: Configuring & Uploading MOH file. Go to Media Resource > MOH Audio File Management and select Upload. To Upload a .wav file, choose from the appropriate location.

Go to Media Resource > Music On Hold Audio Source add new

MOH Audio Stream Number: 2

MOH Audio Source File: and enable activated check box.

Step 2: Configuring phones for MOH.Go to Device > Phone, find 1004 and Select User MOH & associate it with the newly defined MOH stream. Verification: Place a call from 1004 to 1005 and put 1004 on hold. The user on extension1005 will hear newly uploaded audio.

Experiment (iv): Multicast MOH

Aim: To configure multicast MOH in order to save on Bandwidth consumed by the MoH. This will be significant when Simultaneous MoH consumes precious Bandwidth.

Configuration Steps:Step 1: Enabling Multicast MoH. Go to Media Resource > Music On Hold Audio Source, find and enable Allow Multicasting check box in audio stream 1&2.

Go to Media Resource > Music On Hold Audio server, find MOH_2 and check Enable Multicast Audio Source On This MOH Server check box, enter the following details,

Base Multicast IP Address: 239.1.1.1Increment Multicast on parameter: IP Address

Max Hops: 2

Step 2: Configure MRG & MRGL.

Go to Media Resources > Media Resource Group, add new MRG named MOH_MRG , select CFB_2,MOH_2,ANN_2 and check Use Multicast for MOH Audio check box. Go to Media Resources > Media Resource Group List, add new MRGL named MOH_MRGL and select MOH_MRG.

Go to System > Device Pool, Define the newly created MRGL MOH_MRGL to the default Device Pool.

Verification:

Go to Media Resource > Music On Hold Audio server, find MOH_2 and change max hops to 1. Place a call from 1004 (HQ) to 1008 (BO) and put 1004 on hold. No MoH Audio is heard at the far end as the hop count ensures that the Multicast Audio doesnt reach the Branch Office.

Go to Media Resource > Music On Hold Audio server, find MOH_2 and change max hops to 2. Now place a call from 1004 to 1008 and put 1004 on hold. The MoH Audio is now heard.

To verify that the Multicast stream is travelling across both the Voice Gateways, issue the below command on the HQ Voice Gateway & observe that the Outgoing Interface List is NULL when the phone is not on Hold & shows an interface whenever the phone is on HOLD.

R2# show ip mroute (phone not on Hold)(*, 239.1.1.1), 00:01:08/stopped, RP 0.0.0.0, flags: D

Incoming interface: Null, RPF nbr 0.0.0.0

Outgoing interface list:

FastEthernet0/1, Forward/Sparse-Dense, 00:01:08/00:00:00

(10.0.0.10, 239.1.1.1), 00:00:56/00:02:55, flags: PT

Incoming interface: FastEthernet0/1, RPF nbr 60.0.0.3

Outgoing interface list: Null

(*, 224.0.1.40), 00:01:24/00:02:46, RP 0.0.0.0, flags: DCL



FastEthernet0/1, Forward/Sparse-Dense, 00:01:24/00:00:00R2# show ip mroute (Phone on Hold)(*, 239.1.1.1), 00:00:16/stopped, RP 0.0.0.0, flags: DC





(10.0.0.10, 239.1.1.1), 00:00:04/00:02:57, flags: T

Incoming interface: FastEthernet0/1, RPF nbr 60.0.0.3



(*, 224.0.1.40), 00:00:31/00:02:41, RP 0.0.0.0, flags: DCL




This confirms that the Multicast stream is being forwarded on interface Fastethernet 0/0.Experiment (v): Hardware Conference Bridge

Aim: To configure Hardware Conference Bridge using the DSPs of the Gateway & control them from the CUCM. Allow access to this Hardware Conference bridge using Media Resource Group (MRG) & Media Resource Group List (MRGL).Step 1: Configure DSPfarm in R2 CLI.voice-card 0

dspfarm

dsp services dspfarm

!

sccp local loopback0

sccp ccm 10.0.010 identifier 1 version 5.0.1

sccp

!

sccp ccm group 1

associate ccm 1 priority 1

associate profile 1 register HQ- CFB

!

dspfarm profile 1 conference

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

maximum sessions 1

associate application SCCP

no shutdown

Step 2: Configure Hardware Conference Bridge in CUCM.Go to Media Resources > Conference Bridge and click Add New to create a conference bridge with these parameters:

Type: Cisco IOS Enhanced Conference Bridge

Name: HQ-CFB

Device pool: DefaultDevice security mode: Non Secure Conference Bridge, Click Save.

Step 3:Configure Media Resource Group & List.

Go to Media Resources > Media Resource Group, find MOH_MRG, add newly created hardware conference bridge HQ-CFB and remove the software conference bridge CFB_2. Since we have already associated it with device pool and MRGL in the previous experiment, we just have to reset all the phones.

Verification: Now establish a conference between 1004, 1005 & 1006 as before and issue the below command on the Gateway to verify the DSP usage.R2#show dspfarm dsp activeSLOT DSP VERSION STATUS CHNL USE TYPE RSC_ID BRIDGE_ID PKTS_TXED PKTS_RXED

0 1 23.8.3 UP 1 USED conf 1 0x1 316 317

0 1 23.8.3 UP 1 USED conf 1 0x2 316 316

0 1 23.8.3 UP 1 USED conf 1 0x3 316 317

Total number of DSPFARM DSP channel(s) 1.

LAB 7:User Features

Experiment (i): Basic featuresAim : To configure the below user features in CUCM

a. call park b. call pickup group c. DND d. Call Back e. Privacy

Task 1: Call ParkUsing this feature, a call can be temporarily parked (a specific DN number is configured for this) from one extension & can be picked up from a different extension.Configuration: Go to Call Routing > Call Park and add new park number 1111Verification : Now place a call from extension 1001 to 1002. Answer the call from 1002 and press the park softkey, the phone will display as To Park Number: 1111. Hang up 1002. Now dial 1111 from a different extension, say 1003 and to pick up & continue the call.Task 2: Call PickupAn incoming call on one phone extension can be received from another using the Call Pickup feature. For this feature, well have to group several phone extensions together & configure the pick up soft key in the phones using softkey template. A visual alert is seen on all group phones whenever any phone in the group receives an incoming call.

(i) Configure pickup group : Give it a name & associate it with a DN numberGo to Call Routing > Call Pickup Group and add a new group as follows,

Call Pickup Group Name: GRP1

Call Pickup Group Number: 1100

Description: Grp 1

Call Pickup Group Notification Policy: Visual Alert(ii) Configure PickUp Softkey in a newly created softkey template Go to Device > Device Settings > Softkey Template and add new Standard User Template named STD TEMP.

Select Configure Softkey Layout from the drop down box on the right top and select go, add Pick Up softkey in On Hook state.

(iii) Configure Softkey Template & pick up groups to phones Go to Device > Phone and apply the newly created softkey template to the device. Go to the Line Settings and apply the callpickup Group.

Configure phones 1001, 1002 and 1003 with the above 2 steps & they now belong to the same group.Verification:Place a call from 1001 to 1002. Do not answer. Observe the visual alert on 1003. Press the PickUp button on 1003 to answer the call.

Task 3: Configuring Do Not Disturb This feature helps us to place the phone in silent mode where only a visual alert is seen whenever a call is received.

Configuration: Go to Device > Device Settings > Softkey Template and find STD TEMP softkey and add Toggle Do Not Disturb softkey in OnHook, OffHook, RingIn state. Choose this softkey template for all the phones & Restart the phones.Verification: Place a call from 1001 to 1002 do not answer and press DND softkey in 1002, the phone will stop ringing & visual alert on 1002 is seen.

Press More softkey in 1001 and press DND softkey. The phone will display Do Not Disturb is active. Now place a call from 1002 to 1001 & the phone will indicate the incoming call with a short beep and visual alert.

Task 4: Configuring CallBack this feature allows us to automatically redial a number (that is busy) so that the call is placed once the far end phone becomes free. Go to Device > Device Settings > Softkey Template and find STD TEMP softkey and add Call Back softkey on hook,RingOut state. Choose this template for all the phones & restart the phones.Verification: Place a call from 1001 to 1002 and answer the call. When this call is in progress, place a new call from 1003 to 1001 without disconnecting the previous call and press call back softkey on 1003. (You can press CallBack Softkey to verify the status or press cancel to cancel the call back feature). Now disconnect the call between 1001 & 1002 you will get a visual alert on the 1003, press call softkey to call 1001.

Task 5: Configuring Barge (using the IP Phones Built-in conference Bridge).This feature allows a user to barge in to a call in progress. Two phones must be configured with a shared line & Barge feature is to be enabled.

Go to Device > Phone and add a shared line DN 11 on line 2 of 1001 & 1002.

Go to System > Service Parameters , set Privacy Setting to False, set Builtin Bridge Enable to On and check if Party Entrance Tone parameter is True.Verification:

Place a call from (line 2) 1001 to 1003. When this call is in progress, access line 2 of 1002 and press Barge Softkey. Now all the phones display To Barge & all 3 phones are in conference.

Configure Privacy (associated with Barge).

This feature is used to block the Barge capabilities of a shared line whenever any confidential call has to be made. This feature is enabled by adding the Privacy button as below. Go to Device > Device Settings > Phone Button Template and click find Copy Standard 7961 SCCP (or Standard CIPC SCCP, as required) and rename it to Conf_Phone. Change button3 to Privacy. Go Device > Phone 1001 & apply the newly created phone button template and reset the phone.

Verification:Press Privacy button in 1001 and place a call from line 2 of 1001 to 1003. When this call is in progress, access line 2 of 1002 and observe that the Barge Softkey doesnt appear.Experiment (ii): Configuring Presence & Speed Dials

Task 1: Presence enabled Speed Dials.Aim: To configure Presence enabled Speed Dials and Monitor the On Hook / Off Hook status of one phone from another (Presence feature).

Step1: Configuring Phone Button Template. Go to Device > Device Setting > Phone Button Template, find and select Standard 7940 SCCP, copy the template and change its name to 7940 New and configure line 2 as Speed Dial BLF. Go to Device > Device Setting > Phone Button Template, find and select Standard 7961 SCCP, copy the template and change its name 7961 New and configure line 2 & 3 to Speed Dial BLF. Go to Device > Device Setting > Phone Button Template, find and select Standard CIPC SCCP, copy the template and change its name CIPC New and configure line 2 & 3 to SPEED Dial BLF.Step2: Configuring Phones. Go to Device > Phone and apply Specific Phone button templates to respective phones.

Go to Device > Phone, find 1001 and configure Button 2 of 1001 to Monitor 1002.

Go to Device > Phone, find 1002 and configure Button 2 & 3 to Monitor 1003 & 1004. Go to Device > Phone, find 1003 and configure Button 2 & 3 to Monitor 1004 & 1001. Go to Device > Phone, find 1004 and configure Button 2 & 3 to Monitor 1002 & 1003.

Verification: Place a call from 1001 to 1002 and check the presence (on hook/off hook status of 1002) from 1004.

Similarly check for other conditions too.Task 2: Presence Enabled Call Lists.Aim: Enable presence in the call lists, so that if a call is missed, the person can call back after verifying whether the other phone is on hook or off hook (Missed call list).

Step 1: Enabling Presence enabled call lists.Go to System > Enterprise Parameters and enable BLF for Call Lists.Verification:Place a call from 1002 to 1003. Hang up so that a missed call is recorded on phone 1003. Now check the call list of 1003. Presence information of 1002 can be seen in the missed call list.

Note: 7940 does not support Presence enabled Call List.

Task 3: Configuring Presence Policies Aim: To implement presence policies which is as defined below. We use Subscribe CSS & Presence Groups to implement the same.

Tasks:1) MD (1004) can monitor Manager (1003).

2) Manager (1003) can monitor MD (1004).

3) Receptionist (1002) can monitor Manager (1003).

4) Lobby (1001) can monitor Receptionist (1002).

Step 1: Configure Partition & CSS.Configure Partition & CSS as shown below,

MDManagerReceptionistLobby

Manager_CSSXX

Lobby_CSSXX

MD_CSSXX

Rec_CSSXX

Step 2: Configure Presence Groups. Go to System > Presence Group, add new presence groups Lobby, Rec, Man, MD.

Go to System > Presence Group, find and configure presence group policies as follows

MD to Man permitted.

Man to MD permitted.

Rec to Man permitted.

Lobby to Rec permitted.

Go to System > Service Parameters, check Default Inter-Presence Subscription is set to Disallow Subscription.Step 3: Configure Phones. Go to Device > Phone, find 1001 and configure Subscribe CSS as Lobby_CSS, Presence Group as Lobby. Go to Line 1 of 1001 and apply Presence Group as Lobby.Subscribe CSSDevice Presence GrpLine Presence Grp

1001Lobby_CSSLobbyLobby

1002Rec_CSSRecRec

1003Manager_CSSManMan

1004MD_CSSMDMD

Note: Subscribe CSS is similar to CSS but is required to view Presence information.

Device Presence Group is will monitor Line Presence Group.Verification: Check that from 1004 we cant view the presence of 1002.

Similarly from Call Lists check if you can view the presence of 1002

Check if the other conditions are met.Lab 8: Voicemail Integration

Aim: To integrate CUCM with Unity to provide voice mail feature.

Flow chart:

CUCM Configuration

Step1: Configure voice mail ports on CUCM.

In this step we will configure Voice mail ports in CUCM to communicate with Unity.

Go to Voice Mail > Cisco Voice Mail Pilot and add new port as follows

Port Name: CiscoUM1-VI1Device Pool: DefaultDevice Security Mode: Non Secure Voice Mail PortDirectory Number: 9991Port Name: CiscoUM1-VI2Device Pool: DefaultDevice Security Mode: Non Secure Voice Mail PortDirectory Number: 9992Step 2: Configure Message Wait Indicator

MWI causes a lamp on the IP Phone to glow - indicating the user if he has any new messages. Two numbers are to be configured for MWI one for indicating ON condition & another for OFF.

Go to Voice Mail > Message Wait Indicator and click Add New MWI as follows,

Message Waiting Number: 9901Description: MWI OnMessage Wait Indicator: ONMessage Waiting Number: 9902Description: MWI Off

Message Wait Indicator: Off

Step 3: Configure Voice Mail Pilot, Line Group, Hunt List & Hunt Pilot

Voice Mail Pilot is the CTI Route Point to access Voice Mail. A Hunt Pilot with the same number is created & mapped to the Voicemail Pilot. The Hunt Pilot is also mapped to a Hunt List which inturn maps to a Line Group that provides access to the CTI Ports the Voice Mail Ports (which are configured as members of the Line Group). Go to Call Routing > Route/Hunt > Line Group and create a new line group with following details

Line Group Name: VMLGRNA Reversion Tine :5Distribution Algorithm: Top Down, Select 9991 & 9992 from available DN and Add to Line Group.

Go to Call Routing > Route/Hunt > Hunt List and add new hunt list with name VMHL1,select CUCM Group as Default, check the Enable this Hunt List & For Voice Mail Usage option and add line group VMLG by clicking Add Line Group button.

Go to Call Routing > Route/Hunt > Hunt Pilot and add new pilot as follows

Hunt pilot: 9999Hunt List: VMHL1 Go to Voice Mail > Voice Mail Pilot and add new Voice Mail Pilot Number as 9999Step 4: Configure Voice Mail ProfileGo to Voice Mail > Voice Mail Profile and add new voice mail profile as follows,

Voice Mail Profile Name: VMProfile

Voice Mail Pilot: 9999,check the Make this the Default Voice Mail Profile for system Check Box.

Step 5: Configure phone for forwarding to voice mailGo to Devices > Phone and click find and select 1001,go to the Line settings of the phone and check Voice mail check Box for Forward Busy Internal, External, No Answer Internal, external options.Configure Unity for integration with CUCMStep 1 :

Go to Start > Programs > Unity > Manage Integration (Delete any existing integrations), click Create Integration and select Cisco Call Manager - click Next,

configure Integration Name as CUCM1 - click next

enter CUCM ip address & check the connectivity by pressing the ping server button. If successful, click next

skip secondary server configuration

set MWI on & off as 9901,9902 respectively and click next

Enter the number of ports as 2 & click verify button to check the configuration and click next (skip Trunk configuration) Finish.

Reopen the Integration window.

Step 2: Create Voice mail for Users

Create Voice Mail users & associate them to phone extensions

Go to System Administration shortcut on Desktop and Click Subscriber tab, click add button(+ icon) and enter the following details,

First Name: UserLast name: 1Display name: User1Extension: 1001 and click Add button, select Phone Password tab and check password never expires box and enter the password

First Name: UserLast name: 2Display name: User2Extension: 1002Now, click the messages button on the IP phones and enroll the phone for listing by configuring your name and welcome greetings messages.Verification:

Place a call from 1001 to 1002 and dont answer the call. It will forward you to voice mail box of 1002. Record a message and hang up the call. Observe that the light on IP Phone 1002s handset glows - indicating a message in the Voicemail Box. Use the messages button on 1002 to listen to the message. Hang up & the light on 1002s handset goes off.10.0.0.10

1001

1011

1001

1002

T1

PSTN

E1

PSTN

1001

1002

2341005432

E1

PSTN

1001

1002

2341005432

E1

PSTN

1001

1002

2341005432

E1

PSTN

1001

1002

1003

1004

1005

1006

E1

PSTN

1001

1002

2341005432

1003

1004

1005

1005

1006

1004

1007

1008

1001

1002

1003

1001

1002

1003

1004

Configure MWI in CUCM.

3) Configure Voice mail Pilot ,Hunt Pilot, Line Group, Hunt List in CUCM.

Configure Voice Mail Ports in CUCM.

4) Configure Voice Mail Profile in CUCM.

5) Configure Phones for Forwarding to Voice Mail in CUCM.

6) Create Integration in Unity .

Create Voice Mail boxes in Unity.

i

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