Implementing Cisco Unified Communications IP Telephony...

CIPT2

Implementing Cisco Unified Communications IP Telephony Part 2 Version 6.0

NIL Lab Guide

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© 2008, NIL Data Communications Table of Contents I

Table Of Contents

Implementing Basic Multisite Connections 1 1. Objective 1 2. Lab Topology 2 3. Addressing and Routing 4 4. Detailed Instructions 5

Implementing Multisite Dial Plans 9 1. Objective 9 2. Command List 10 3. Lab Topology 11 4. Addressing and Routing 13 5. Detailed Instructions 14

Implementing SRST and MGCP Fallback 39 1. Objective 39 2. Lab Topology 40 3. Addressing and Routing 42 4. Detailed Instructions 43

Implementing Cisco Unified Communicma boliations Manager Express as SRST Fallback 51 1. Objective 51 2. Lab Topology 52 3. Addressing and Routing 54 4. Detailed Instructions 55

Implementing Bandwidth Management 61 1. Objective 61 2. Lab Topology 62 3. Addressing and Routing 64 4. Detailed Instructions 65

Implementing Call Admission Control 77 1. Objective 77 2. Lab Topology 78 3. Addressing and Routing 80 4. Detailed Instructions 81

Implementing Device Mobility 91 1. Objective 91 2. Lab Topology 92 3. Addressing and Routing 94 4. Detailed Instructions 95

Implementing Extension Mobility 99 1. Objective 99 2. Lab Topology 100 3. Addressing and Routing 102 4. Detailed Instructions 103

Implementing Cisco Unified Mobility 109 1. Objective 109 2. Lab Topology 110 3. Addressing and Routing 112 4. Detailed Instructions 113

Implementing Security in Cisco Unified Communications Manager 123 1. Objective 123 2. Lab Topology 124 3. Addressing and Routing 126 4. Detailed Instructions 127

II Table of Contents © 2008, NIL Data Communications

© 2008, NIL Data Communications NIL Lab Guide 1

CIPT260

Implementing Basic Multisite Connections

1. Objective In this exercise you will add an H.323 and a MGCP gateway to the Cisco Unified Communications Manager (CUCM) and configure IOS routers to act as H.323 and MGCP gateways. You will add an H.225 intercluster trunk to Cisco Unified Communications Manager and configure it as a gatekeeper-controlled trunk. Finally, you will create in the CUCM a SIP trunk to an IP Telephony Service Provider (ITSP).

2 Implementing Cisco Unified Communications IP Telephony Part 2 (CIPT2) v6.0 © 2008, NIL Data Communications

2. Lab Topology Devices in the lab are connected according to the setup in Figure 2-1.

Figure 2-1: Lab topology

Note Classroom phones have precedence over CIPC-emulated phones. Use CIPC-emulated phones Phonex-1 to Phonex-3 only if you have no classroom phones. Otherwise, ignore the CIPC-emulated phones.


Each pod includes one Cisco Unified Communications Manager, four PC-emulated phones (including a PSTN phone), and three routers (including a PSTN router). Additionally, there is one gatekeeper router per workgroup (two pods). If you use classroom equipment, you will also have one classroom router, one classroom switch and three IP phones per pod, as detailed in Table 2-1 (x is the pod number).

Device name Device role in the laboratory

CUCMx-1 Cisco Unified Communications Manager node

HQx Voice gateway router acting as a H.323 gateway to the PSTN via a digital E1 trunk.

BRx Voice gateway router acting as a MGCP gateway to the PSTN via a digital E1 trunk.

PSTN-Routerx PSTN voice gateway router connected to HQx and BRx providing connectivity to the PSTN network in the lab.

GK Router acting as a gatekeeper for the intercluster trunk, and a proxy of a SIP provider.

PSTN-Phonex PC running Cisco IP Communicator (CIPC) simulating a PSTN phone.

Phonex-1, Phonex-2, Phonex-3

PCs running Cisco IP Communicator (CIPC) that are used if no classroom phones are available.

Classroom Router Router provides Voice and Data VLAN connectivity from classroom equipment to the remote equipment. Only necessary in the security lab.

Classroom Switch Switch into which classroom IP phones are plugged. Only necessary is the security lab.

Classroom phones Phonex-1, Phonex-2, Phonex-3

If available, should be used instead of the CIPC-emulated phones Phone1, Phone2, and Phone3. Necessary in the security lab.

Table 2-1: Roles of devices in the lab (x is the pod number)

User Credentials Information Use the following credentials to log in to the lab devices, which require authentication.

System Username/password

CIPC-emulated Phones Administrator/admin

Cisco Unified Communications Manager Administration

CCMAdministrator/appuserpass

Cisco Unified Communications Serviceability


Cisco Unified OS Administration admin/adminpass

Table 2-2: User credentials information


3. Addressing and Routing This section contains information on IP addressing used in the initial configuration of the lab.

IP Addressing Scheme Table 3-1 lists the networks used in this lab (x is your pod number).

Parameter Value

CUCM VLAN 10.x.1.0/24

HQ classroom Voice VLAN (classroom IP phones Phonex-1 and Phonex-2)

10.x.5.0/24

HQ CIPC Voice VLAN (CIPC-emulated phones Phonex-1 and Phonex-2)

10.x.4.0/24

HQ classroom Data VLAN (PC plugged into classroom phone) 10.x.3.0/24

BR classroom Voice VLAN (classroom IP phone Phonex-3) 10.x.5.0/24

BR CIPC Voice VLAN (CIPC-emulated phone Phonex-3) 10.x.6.0/24

PSTN LAN 10.3.0.0/24

Table 3-1: IP networks used in the lab exercise (x is the pod number)

The individual addresses assigned to network interfaces are displayed in Figure 3-1.

Figure 3-1: IP address assignment in the lab exercise


4. Detailed Instructions Follow the steps in the tasks to implement basic multisite connections.

Task 1: Add an H.323 Gateway to Cisco Unified Communications Manager

In this task you will add the HQ H.323 gateway to Cisco Unified Communications Manager.

Note The character x denotes your pod number.

Step 1 Connect to your Phonex-1 (x is your pod number) and login in as administrator with password admin. Access CUCM Administration by clicking its desktop shortcut (https://10.x.1.1/ccmservice). Log in as CCMAdministrator with password appuserpass. Update the MAC addresses of your classroom phones, if needed.

Note Alternatively you could also access CUCM Administration/Serviceability from a PC plugged to Phonex-1 in the classroom.

Step 2 Navigate to Device > Gateway and click the Add New button. Step 3 Select H.323 Gateway, click Next and configure these parameters:

Device Name: 192.168.x.1 (x is your pod number) Description: HQ PSTN Gateway (H.323) Device Pool: Default

Step 4 Leave all other parameters to the default settings, click Save and then OK.

Verification Step 5 In CUCM Administration choose Device > Gateway and click Find. Verify that gateway is

listed and verify its configuration.

Task 2: Configure an H.323 Gateway In this task you will configure an IOS H.323 gateway for connecting to the PSTN at the HQ.

Step 6 Connect to your HQ router by clicking its icon in the lab topology. Step 7 Set the LAN interface that connects to your CUCM as the source interface for H.323 packets (x

is your pod number): interface Loopback 0 h323-gateway voip interface h323-gateway voip bind srcaddr 192.168.x.1

Configuration 4-1: Set the LAN interface

Note Additional configuration and verification will be done in the next lab activity.


Task 3: Add an MGCP Gateway to Cisco Unified Communications Manager

In this task you will add an MGCP gateway to Cisco Unified Communications Manager.

Step 8 Choose Device > Gateway and click the Add New button. Step 9 Choose the gateway platform Cisco 2811 and click Next. Step 10 Select MGCP as the protocol, click Next, and configure these parameters:

Domain Name: BRx (must match the hostname of the router and is case sensitive, x is your pod number)

Description: BR MGCP gateway Cisco Unified Communications Manager Group: Default Module in Slot 0: NM-4VWIC-MBRD (module with 2 E1 ports) Global ISDN Switchtype: EURO

Step 11 Click Save and then select the module VWIC-2MFT-E1 as the Subunit 2 in Slot 0. Click Save. Step 12 Click the port icon 0/2/0 (the left endpoint with the question mark). Step 13 Set the Device Pool to Default and Channel Selection Order to Top Down.

Note TOP_DOWN means first to last, in this case 1 to 32. BOTTOM_UP means last to first, in this case 32 to 1.

Step 14 View the remaining parameters without changing them and save.

Verification Step 15 In CUCM Administration choose Device > Gateway and click Find. Verify that gateway is

listed and verify its configuration

Task 4: Configure an MGCP gateway In this task you will configure a Cisco IOS MGCP PSTN gateway located at the Branch to register with Cisco Unified Communications Manager.

Step 16 Access your BRx router by clicking its icon in the lab topology. Step 17 Debug configuration server feature events using the debug ccm-manager config-download

events command. Step 18 Enter in the global configuration mode the following commands:

ccm-manager config server 10.x.1.1 ccm-manager config

Configuration 4-2: debug configuration server

Note The gateway will now pull its MGCP configuration from the Cisco Unified Communications Manager TFTP server.


Step 19 Watch the debug output to monitor the operation of the configuration server feature. Turn off all debug by entering no debug all.

Step 20 View the router configuration. Your gateway should be configured for MGCP. The configuration that was added by the configuration server feature includes MGCP, controller and ISDN PRI settings.

Verification Step 21 On your BR router, verify the mgcp registration status using the commands show ccm-manager

hosts (should be ‘registered’), show mgcp endpoint (all controlled ISDN PRI endpoint ports should be up), and show mgcp (Admin State and the Oper State should be active).

Step 22 In CUCM administration navigate to Device > Gateway. Select the option to show endpoints and click Find. The Status of the endpoint should be ‘Registered with 10.x.1.1’.

Note Further verification will be done in the next lab exercise.

Task 5: Configure an H.225 Trunk in Cisco Unified Communications Manager

In this task you will add an H.225 intercluster trunk to Cisco Unified Communications Manager, which will be gatekeeper controlled. The gatekeeper runs on the PSTN router.

Step 23 Navigate to Device > Gatekeeper, click Add New and configure these gatekeeper parameters: Host Name/IP Address: 192.168.3.1 Description: Gatekeeper

Step 24 Click Save, and then reset the newly added gatekeeper. Step 25 Choose Device > Trunk, click Add New and choose H.225 Trunk (Gatekeeper Controlled).

Click Next. Step 26 Configure a trunk named H225-Trunk-x (x is your pod number) with these parameters:

Select Device Pool Default. In the Gatekeeper Information pane:

— Gatekeeper Name: 192.168.3.1 — Terminal Type: Gateway — Technology Prefix: 1#* — Zone: podx

Note The zone name is case sensitive.

Step 27 Click Save, and then click OK on the pop-up window.


Verification Step 28 Connect to the GK router by either clicking its icon in the lab topology or telnet-ing to it

(192.168.3.1) from any device. Step 29 View the registered endpoints using the show gatekeeper endpoints. The IP address of your

CUCM should be listed with an H323-ID of H225-Trunk-x.

Note Further verification will be done in the next lab exercise.

Task 6: Configure a SIP trunk in Cisco Unified Communications Manager

In this task, you will add a SIP trunk to an IP Telephony Service Provider (ITSP). The ITSP functionality is provided by the GK and PSTN routers.

Step 30 In CUCM Administration, choose Device > Trunk and click Add New. Step 31 Choose SIP Trunk, click Next and configure these parameters:

Device Name: SIP_Trunk Description: Trunk to ITSP Device Pool: Default

Step 32 In the SIP Information pane enter the IP address of the ITSP in the Destination Address field: 192.168.3.1.

Step 33 Make sure that Destination Address is an SRV box is not selected. Step 34 Choose Non Secure SIP Trunk Profile from the SIP Trunk Security Profile and Standard SIP

Profile from the SIP Profile menu. Step 35 Click Save, and then OK.

Note The next exercise (“Implementing Multisite Dialplans”) includes a verification of the tasks you configured in this exercise. You can directly proceed to the next exercise without asking the instructor to load it. If the instructor loads the next exercise, your configuration will be overwritten by the preconfigured settings equivalent to what you configured in this lab exercise.


CIPT260

Implementing Multisite Dial Plans

1. Objective In this exercise you will implement a dial plan to support inbound and outbound PSTN calls, site-code dialing, tail-end hop-off, and PSTN backup. To achieve that goal, you will configure partitions, calling search spaces (CSS's) and assign partitions and calling search spaces to phones and phone directory numbers. You will configure the voice gateways for incoming PSTN calls by creating the required translation profiles and dial peers at the Cisco IOS H.323 gateway, assigning the appropriate CSS to the gateways and setting the significant digits to four. Next, you will configure PSTN route patterns that point to route lists. The route lists point to route groups that contain the voice gateways. You will set up appropriate digit manipulation for calling and called party numbers at the gateways and in the CUCM. Then you will update the route lists to allow the use of a remote gateway in case of failure of the preferred local gateway. You will configure a dial plan that allows intersite calls to use the H.225 trunk. Digit manipulation in Cisco Unified Communications Manager should ensure correct presentation of called numbers through the trunk. Next, you will update the route lists to allow the use of a local gateway in case of failure of the preferred H.225 trunk. You will configure a dial plan that allows Tail-End Hop Off (TEHO) for calls to specific PSTN area codes. Finally, you will configure a dial plan for calls which are prefixed with a carrier selection code indicating that the call should be sent via an ITSP. You change the SIP trunk configuration to point to a gateway instead of directly pointing to the ITSP. The Cisco Unified Border Element will act as a signaling and media proxy towards the ITSP. This way, the IP addresses of Cisco Unified Communications Manager and all IP phones can be hidden from the outside. Only the Cisco Unified Border Element needs to have connectivity to the ITSP.


2. Command List The table lists the Cisco IOS commands are used in this exercise.

Command Description

debug isdn q931 Displays ISDN Q931 signaling information

debug voice ccapi inout Debugs the call signaling

debug voip dialpeer inout | all Debugs voice dial peers

show ccm-manager Displays the status of the Cisco Unified Communications Manager registration

show dialplan number Displays which outgoing dial peer is reached when a particular telephone number is dialed

show mgcp Displays the status of MGCP

show mgcp endpoint Displays a list of MGCP endpoints

show voice call summary Displays the list of active calls and their parameters

Table 2-1: IOS commands used in the lab exercise

































Parameter Value



10.x.5.0/24


10.x.4.0/24




PSTN LAN 10.3.0.0/24





5. Detailed Instructions Follow the steps in the tasks to implement multisite dial plans.

Task 1: Configure Partitions and Calling Search Spaces In this task, you will configure all partitions and calling search spaces (CSS) which are required for the following tasks. You will assign partitions and calling search spaces to phones and phone directory numbers.

Step 1 Connect to your Phonex-1 (x is your pod number) and login in as administrator with password admin. Access CUCM Administration by clicking its desktop shortcut (https://10.x.1.1/ccmservice). Log in as CCMAdministrator with password appuserpass. Update the MAC addresses of your classroom phones as needed.


Step 2 In CUCM Administration go to Call Routing > Class of Control > Partition, and click Add New.

Step 3 Enter these partition names and their descriptions using this format: HQ_Phones, Headquarters IP Phones BR_Phones, Branch IP Phones HQ_PSTN, PSTN Access (HQ) BR_PSTN, PSTN Access (BR) HQ_ICT, Intercluster Access (HQ) BR_ICT, Intercluster Access (BR) HQ_SIP, ITSP Access (HQ) BR_SIP, ITSP Access (BR) ICT_IN, Internal and TEHO Access for incoming ICT Calls

Configuration 5-1: Partition names

Step 4 Click Save. Step 5 Go to Call Routing > Class of Control > Calling Search Space, and click Add New. Step 6 Create CSS’s and add to them partitions detailed below:

CSS Name Partitions in the CSS

HQ-Phones_css HQ_Phones, BR_Phones, HQ_PSTN, HQ_ICT, HQ_SIP

BR-Phones_css BR_Phones, HQ_Phones, BR_PSTN, BR_ICT, BR_SIP

Inbound-PSTN_css HQ_Phones, BR_Phones

Inbound-ICT_css HQ_Phones, BR_Phones, ICT_IN

Table 5-1: CSS settings

Note Use the Shift key to highlight multiple contiguous entries and the Ctrl key to select multiple non-contiguous entries.

Step 7 Go to Device > Phone and click the Find button.

Note Classroom phones have precedence over CIPC-emulated phones. Use CIPC-emulated phones Phone1 to Phone3 only if you have no classroom phones. Otherwise, ignore the CIPC-emulated phones.


Step 8 For the phones Phonex-1 to Phonex-3, enter the configuration of their first line (Line [1]) and set the partition. From the Related Links select Configure Device and click Go to return to the device level and set the CSS’s as detailed below. If you use classroom phones, set their MAC addresses to the real values.

Note If you have classroom phones, configure the phones with symbols 79XX. If you do not use classroom phones, configure the entries with CIPC symbols.

Phone line Partition of phone line CSS of phone

Phonex-1: Line [1] - 2001 HQ-Phones HQ-Phones_css

Phonex-2: Line [1] - 2002 HQ-Phones HQ-Phones_css

Phonex-3: Line [1] - 2003 BR-Phones BR-Phones_css

Table 5-2: Phone line settings

Note Make sure that you assign the CSS to the phone at the device level (phone configuration page): Do not assign the CSS at the line level.

Step 9 Go to Device > Phone and click the Find button. Select all phones and click Reset Selected, then Reset and Close.

Verification Step 10 Each phone should still be able to call the other two phones in the pod. Verify this by placing test

calls.

Note If you use CIPC-emulated phones instead of classroom IP phones and want to hear sounds, follow the procedure in the appendix. Whenever you reset a CIPC phone, you must first terminate the RDP session with sound and wait until the CIPC registers with the ‘w/o sound’ option.


Task 2: Configure Inbound PSTN Calls In this task, you will configure the gateways for incoming PSTN calls. You will configure the required translation profiles and dial peers at the Cisco IOS H.323 gateway (HQ). Then you will assign the appropriate CSS to the HQ gateway in Cisco Unified Communications Manager. For the MGCP gateway used in the branch, you will assign the appropriate CSS and set the significant digits to four.

Configure a Voice Translation Profile at the HQ H.323 Gateway

In these steps you will configure a voice translation profile to manipulate the called party number of PSTN calls received at the ISDN interface of the HQ router. The called E.164 number (51x-555-2XXX or 555-2XXX) should be translated to four-digit internal directory numbers (2XXX). In addition, at the HQ gateway, 9011 will be prefixed to incoming international calling party numbers, 91 will be prefixed to incoming long distance calling party numbers, and 9 will be prefixed to incoming local calling party numbers.

Step 11 Connect to your HQ router by clicking its icon in the lab topology. Step 12 Configure a voice translation profile pstn-in which translates the 10 digit and 7 digit called party

PSTN number of the HQ phones to their internal 4 digit directory number. It should prefix the digits that are required when placing outgoing calls to the calling party. (Replace x by your pod number).

voice translation-rule 1 rule 1 /^51x5552/ /2/ rule 2 /^5552/ /2/ ! voice translation-rule 2 rule 1 /^.*/ /9&/ type subscriber subscriber rule 2 /^.*/ /91&/ type national national rule 3 /^.*/ /9011&/ type international international ! voice translation-profile pstn-in translate called 1 translate calling 2 ! voice-port 0/2/0:15 translation-profile incoming pstn-in

Configuration 5-2: Voice translation profile PSTN

Configure dial peers at the HQ H.323 gateway for calls to headquarters phones

Step 13 On your HQ router, configure dial peers to allow incoming PSTN calls. The incoming pots dial peer should support direct inward dial, the outgoing voip dial peer should point to your CUCM.

! dial-peer voice 1 voip destination-pattern 2... session target ipv4:10.x.1.1 ! dial-peer voice 2 pots direct-inward-dial incoming called-number .

Configuration 5-3: Dial peers


Assign Calling Search Space to the HQ Gateway in CUCM

Step 14 In CUCM Administration select Device > Gateway and click Find. Step 15 Click the 192.168.x.1 Device Name to enter the gateway settings. Step 16 In the Call Routing Information - Inbound Calls pane set the Calling Search Space to Inbound-

PSTN_css Step 17 Click Save and then click OK on the pop-up window. Step 18 Reset the gateway.

Set the significant digits to four and assign CSS to the BR Gateway in CUCM

Step 19 Go to Device > Gateways and click the Find button. Step 20 Click the See Endpoint link and then the available device. Step 21 In the Call Routing Information - Inbound Calls pane set the Significant Digits to 4 and the

Calling Search Space to Inbound-PSTN_css. Step 22 Click Save, then OK and reset the gateway.

Verification Step 23 Place a call from your PSTN phone, line 1 (Local) to your HQ (2001 or 2002) by dialing a full

PSTN number (1-51x-555-2XXX or 555-2XXX). The calling party number should be shown as a 7 digit number with an extra 9 in front.

Step 24 Place a call from your PSTN phone, line 2 (LD) to your HQ by dialing a full PSTN number (1-51x-555-2XXX or 555-2XXX). The calling party number should be shown as a 10 digit number with an extra 9-1 in front.

Step 25 Place a call from your PSTN phone, line 3 (Intl) to your HQ by dialing a full PSTN number (1-51x-555-2XXX or 555-2XXX). The calling party number should be shown as an international number with 9011 in front.

Step 26 Place a call from your PSTN phone, line 1 (Local) to your branch by dialing a full PSTN number (1-52x-555-3001 or 555-3001). The calling party number should be shown as a 7 digit number (because you did not manipulate the calling number on the MGCP gateway).

Step 27 Place a call from your PSTN phone, line 2 (LD) to your branch by dialing a full PSTN number (1-52x-555-3001 or 555-3001). The calling party number should be shown as a 10 digit number (because you did not manipulate the calling party number on the MGCP gateway).

Step 28 Place a call from your PSTN phone, line 3 (Intl) to your branch by dialing a full PSTN number (1-52x-555-3001 or 555-3001). The calling party number should be shown as an international number (without 9011 in front, because as you did not manipulate the calling party number on the MGCP gateway).


Task 3: Configure Outbound PSTN Calls Using an H.323 Gateway

In this task, you will configure PSTN route patterns that point to a route list. The route list will be configured to point to a route group including the HQ H.323 gateway. Appropriate digit manipulation for calling and called party numbers will be configured at the H.323 gateway.

Step 29 Navigate to Call Routing > Route/Hunt > Route Group and click the Add New button. Step 30 Enter HQ_rg for the Route Group Name. Step 31 From the Available Devices pane add the H.323 gateway (shown with its ip address 192.168.x.1)

by selecting the gateway and clicking Add to Route Group button. Click Save. Step 32 Navigate to Call Routing > Route/Hunt > Route List and click the Add New button. Step 33 Create a route list named HQ-PSTN_rl with description HQ PSTN Access and CUCM group

Default. Click Save. Step 34 Click Add Route Group and select HQ_rg from the list.

Note Digit manipulation will be configured directly at the H.323 gateway.

Step 35 Click Save, then OK and reset the route list. Step 36 Go to Call Routing > Route/Hunt > Route Pattern and click Add New. Step 37 Configure the following route patterns, all in partition HQ_PSTN and pointing to the route list

HQ-PSTN_rl. You may use the Copy button.

Route Pattern Description

9.1800XXXXXXX HQ: PSTN - Toll Free

9.[2-9]XXXXXX HQ: PSTN - Local

9.1[2-9]XX[2-9]XXXXXX HQ: PSTN - Long Distance

9.011! HQ: PSTN - International (with interdigit timeout)

9.011!# HQ: PSTN - International (with #)

911 Activate Urgent Priority Prefix 9

HQ: Emergency (without PSTN access code)

9.911 Activate Urgent Priority

HQ: Emergency (with PSTN access code)

Table 5-3: Route patterns

Note Activate the Urgent Priority check box for the two emergency route patterns (911 and 9.911). At the 911 route pattern configure Prefix Digits (Outgoing Calls) 9 in the Called Party Transformations area. This is required because the H.323 gateway will remove the 9 at the beginning of each PSTN route pattern.

Step 38 Connect to your HQ router and configure a dial-peer: dial-peer voice 2 pots destination-pattern 9T port 0/2/0:15

Configuration 5-4: Dial peer


Note This configuration removes access code 9 from the called number because all matched digits of the destination-pattern command configured at a pots dial-peer are automatically stripped off.

Step 39 On your HQ router, configure a voice translation profile pstn-out which translates the 4 digit calling party number of the HQ phones to their 10 digit E.164 PSTN number.

voice translation-rule 3 rule 1 /^2/ /51x5552/ ! voice translation-profile pstn-out translate calling 3 ! voice-port 0/2/0:15 translation-profile outgoing pstn-out

Configuration 5-5: Voice translation profile

Verification

Note If you want to hear sounds on your PSTN phone, follow the procedure in the appendix.

Step 40 From a HQ phone (2001 or 2002), dial any PSTN number for each configured route pattern: Local – for example: 9-321-4444 LD – for example: 9-1-321-321-4444 Intl – for example: 9011-49-89-4423250 and 9011-49-89-4423250# Toll Free: for example: 9-1-800-123-1234 Emergency – for example: 9-911 and 911

Note Each call should be received at the appropriately labeled line of the PSTN phone. The calling party number should be 51x5552001 or 51x5552002, depending on the used phone.

Step 41 On your HQ router, you can use debug voice translation and debug isdn q931 to verify digit manipulation and ISDN signaling messages. Turn off debugging, when finished.


Task 4: Configure Outbound PSTN Calls Using an MGCP Gateway

In this task, you will configure PSTN route patterns that point to a route list. The route list will be configured to point to a route group including the BR MGCP gateway. Appropriate digit manipulation for calling and called party numbers will be configured in Cisco Unified Communications Manager.

Step 42 Select Call Routing > Route/Hunt > Route Group and click Add New. Step 43 Configure a route group BR_rg pointing to the BRx gateway. Save it. Step 44 Select Call Routing > Route/Hunt > Route List and click Add New. Step 45 Configure a route list with these parameters:

Name: BR-PSTN_rl Cisco Unified Communications Manager Group: Default

Step 46 Click Save and then Add Route Group to add BR_rg from the Route Group drop-down list. Step 47 In the Calling Party Transformations pane enter 52x5553XXX in the Calling Party Transform

Mask field. Step 48 In the Called Party Transformations pane select from the Discard Digits drop-down list

NANP:PreDot.

Note Since BRx is an MGCP gateway, digit manipulation must be configured in Cisco Unified Communications Manager. Digit manipulation is configured at the route list (per route group) and not at the route pattern because in later tasks multiple route groups with different digit manipulation requirements will be configured for the same route list.

Step 49 Click Save and then OK in the pop-up window. Step 50 Go to Call Routing > Route/Hunt > Route Pattern and click Add New. Step 51 Configure route patterns using the details from the table, all in the partition BR_PSTN pointing

to the route list BR-PSTN_rl.

Route Pattern Description

9.1800XXXXXXX BR: PSTN - Toll Free

9.[2-9]XXXXXX BR: PSTN - Local

9.1[2-9]XX[2-9]XXXXXX BR: PSTN - Long Distance

9.011! BR: PSTN - International (with interdigit timeout)

9.011!# BR: PSTN - International (with #)

911 Activate Urgent Priority

BR: Emergency (without PSTN access code)

9.911 Activate Urgent Priority

BR: Emergency (with PSTN access code)

Table 5-4: Route patterns


Note Make sure to activate the Urgent Priority check box for the two emergency route patterns (911 and 9.911). At the 911 route pattern do not configure Prefix Digits (Outgoing Calls) 9 in the Called Party Transformations area. This is not required because the PreDot discard digit instruction configured at the route list for the corresponding route group is ignored for a pattern that does not include a dot.

Verification Step 52 From Phonex-3, dial PSTN numbers for each configured route pattern:

Local – for example: 9-321-4444 LD – for example: 9-1-321-321-4444 Intl – for example: 9011-49-89-12345657 and 9011-49-89-1234567# Toll Free: for example: 9-1-800-123-1234 Emergency – for example: 9-911 and 911

Note Each call should be received at the appropriately labeled line of the PSTN phone. The calling party number should be 52x5553001.

Step 53 At the branch gateway you can use debug isdn q931 to verify ISDN signaling messages.When done, turn off all debugging.

Task 5: Configure PSTN Backup In this task, you will update the route lists HQ-PSTN_rl and BR-PSTN_rl to allow the use of a remote gateway in case of failure of the preferred local gateway.

Step 54 Go to Call Routing > Route/Hunt > Route List, click Find and select HQ-PSTN_rl. Step 55 Click Add Route Group and select BR_rg from the list.

Note As the gateway used by this route group is an MGCP gateway, digit manipulation must be configured in Cisco Unified Communications Manager.

Step 56 In the Calling Party Transformations pane set the Calling Party Transform Mask to 52x5553001 (where x is your pod number). In the Called Party Transformations select NANP:PreDot from the Discard Digits list.

Step 57 Click Save and then OK. Step 58 Make sure that the route group HQ_rg is listed before the BR_rg. Step 59 Go to Call Routing > Route/Hunt > Route List, and select BR-PSTN_rl. Step 60 Click Add Route Group and select HQ_rg from the list.

Note Digit manipulation will be configured directly at the H.323 gateway for calls going through the H.323 gateway.


Step 61 Click Save and then OK in the pop-up window. Step 62 Make sure that the route group BR_rg is listed before the HQ_rg. Step 63 On your HQ router, update the voice translation profile pstn-out to translate the calling number

to 51x5552001 (HQ attendant) when calls arrive from the Branch: voice translation-rule 3 rule 2 /^.*/ /51x5552001/

Configuration 5-6: Update voice translation profile

Note The translation rule has already been assigned to the pstn-out profile, as well as the pstn-out profile, has already been bound to the voice port.

Step 64 On your HQ router, modify the existing dial-peer voice 1 voip to be used as the incoming dial peer for all calls, regardless of the calling party number:

dial-peer voice 1 voip incoming called-number .

Configuration 5-7: Modify the existing dial peer voice

Note Before, this dial peer was selected as incoming dial peer for calls coming from HQ phones only. These calls matched the destination-pattern 2… with their calling number. Now, the calling number is 2XXX or 3XXX; therefore the incoming called-number command is used for incoming dial peer selection.

Verification Step 65 On your HQ and BR routers, activate debug isdn q931. Step 66 On your HQ router, shut down Loopback0 (H323 interface). Step 67 From a HQ Phone (2001 or 2002) dial any PSTN number (e.g. 9-1-333-333-4444). Step 68 The PSTN phone rings, and as calling party number you should see 52x5553001. At the BR

router you should see the appropriate debug output. Step 69 On your HQ router re-enable Loopback0. Step 70 On your BR router, shut down the E1 controller 0/2/0:

controller e1 0/2/0 shutdown

Configuration 5-8: Shut down E1 on your BR router

Step 71 From Phonex-3 dial any PSTN number (e.g. 9-1-333-333-4444). Step 72 The PSTN phone rings, and as calling party number you should see 51x5552001. Check the

appropriate debug output on your HQ router. Step 73 On your BR router reactivate the E1 controller. Disable debugging on both routers.


Task 6: Configure Site Codes for Intercluster Calls from the HQ In this task, you will configure a dial plan that allows intersite calls to use the H.225 trunk. Configure digit manipulation in Cisco Unified Communications Manager to ensure correct presentation of called numbers through the trunk.

Step 74 Select Call Routing > Route/Hunt > Route Group and click Add New. Step 75 Create a route group named H225-Trunk_rg. Add H225-Trunk-x to it and click Save. Step 76 Choose Call Routing > Route/Hunt > Route List and Add New. Step 77 Create a route list named HQ-ICT-HQ_rl with CUCM Group Default and click Save. This

route list will be for calling HQ phones in neighbor pod. Step 78 Click Add Route Group and select H225-Trunk_rg from the list.

Note The H.225 trunk refers to a gatekeeper. The gatekeeper is configured to route calls based on PSTN are codes. Therefore the called number must be transformed to a 10 digit number.

Step 79 In the Calling Party Transformations enter 80x in the Prefix Digits (Outgoing Calls) field (x is your pod number).

Step 80 In the Called Party Transformations select NANP:PreDot from the Discard Digits drop-down list.

Step 81 In the Called Party Transformations pane enter in the Prefix Digits (Outgoing Calls) field 51y555 (y is the neighbor pod number).

Step 82 Click Save and then OK. Step 83 Click Add New. Step 84 Create a route list named HQ-ICT-BR_rl with CUCM Group Default and click Save. This

route list will be for calling BR phones in neighbor pod. Step 85 Click Add Route Group and select H225-Trunk_rg from the list. Step 86 In the Calling Party Transformations pane enter 80x in the Prefix Digits (Outgoing Calls) field (x

is your pod number). Step 87 In the Called Party Transformations pane select NANP:PreDot from the Discard Digits drop-

down list. Step 88 In the Called Party Transformations pane enter in the Prefix Digits (Outgoing Calls) field 52y555

(where y is the pod you connect to). Step 89 Click Save and then OK. Step 90 Choose Call Routing > Route/Hunt > Route Pattern and click Add New. Step 91 Create a route pattern with these parameters:

Route Pattern: 80y.2XXX (where y is your neighbor pod number) Route Partition: HQ_ICT Description: HQ Intersite to neighbor HQ Gateway/Route List: HQ-ICT-HQ_rl

Step 92 Click Save. Click OK in the pop-up window.


Step 93 Click the Copy button and edit these parameters: Route Pattern: 80y.3XXX (where y is the neighbor pod number) Description: HQ Intersite to neighbor BR Gateway/Route List: HQ-ICT-BR_rl

Step 94 Click Save. Click OK in the pop-up window. Step 95 Choose Device > Trunk, click Find and select the H225-Trunk-x. Step 96 In the Inbound Calls pane set the Significant Digits to 4 and the CSS to Inbound-ICT_css. Step 97 Click Save, then OK and reset the trunk.

Verification

Note Before placing calls to the other pod, consult with the students of the other pod. The call can only go through once they finished the configuration.

Step 98 From an HQ phone call a phone in the other pod HQ: dial 80y2001 or 80y2002. Step 99 The other phone should ring. The called IP phone should display a calling party number which

consists of your access and site code (80x) and the directory number of your phone (2001 or 2002).

Note You can verify gatekeeper operation by connecting to it and entering the command debug ras or debug gatekeeper main 5.

Step 100 From an HQ phone call the phone located in the branch of the other pod (dial 80x3001). The called phone should display the calling party number in the format: 80x2XXX.

Task 7: Configure Site Codes for Intercluster Calls from the BR In this task, you will configure a dial plan that allows intersite calls to use the H.225 trunk. Configure digit manipulation in Cisco Unified Communications Manager to ensure correct presentation of called numbers through the trunk.

Step 101 Choose Call Routing > Route/Hunt > Route List and Add New. Step 102 Create a route list named BR-ICT-HQ_rl with CUCM Group Default and click Save. This

route list will be used for calling HQ phones in the neighbor pod. Step 103 Click Add Route Group and select H225-Trunk_rg from the list. Step 104 In the Calling Party Transformations enter 80x in the Prefix Digits (Outgoing Calls) field (x is

your pod number). Step 105 In the Called Party Transformations select NANP:PreDot from the Discard Digits list and

enter 51y555 (y is the neighbor pod number) in the Prefix Digits (Outgoing Calls) field. Step 106 Click Save and then OK. Step 107 Click Add New. Step 108 Create a route list named BR-ICT-BR_rl with CUCM Group Default and click Save. This route

list will be used for calling BR phones in the neighbor pod.


Step 109 Click Add Route Group and select H225-Trunk_rg from the list. Step 110 In the Calling Party Transformations enter 80x in the Prefix Digits (Outgoing Calls) field (x is

your pod number). Step 111 In the Called Party Transformations select NANP:PreDot from the Discard Digits drop-

down list and enter 52y555 (y is the neighbor pod number) in the Prefix Digits (Outgoing Calls) field.

Step 112 Click Save and then OK. Step 113 Choose Call Routing > Route/Hunt > Route Pattern and click Add New. Step 114 Create a route pattern with these parameters:

Route Pattern: 80y.2XXX (where y is your neighbor pod number) Route Partition: BR_ICT Description: BR Intersite to neighbor HQ Gateway/Route List: BR-ICT-HQ_rl

Step 115 Click Save. Click OK in the pop-up window. Step 116 Click the Copy button and edit these parameters:

Route Pattern: 80y.3XXX (where y is the neighbor pod number) Description: BR Intersite to neighbor BR Gateway/Route List: BR-ICT-BR_rl

Step 117 Click Save and then OK.

Verification

Note Before placing calls to the other pod, consult with the students of the other pod. The call can only go through once they finished the configuration.

Step 118 From your branch (Phonex-3) call a phone in the other pod HQ: dial 80y2001 or 80y2002.The called IP phone should display a calling party number which consists of your access and site code (80x) and the directory number of your phone (3001).

Step 119 From your branch (Phonex-3) call the phone in the other pod branch: dial 80y3001.The called IP phone should display a calling party number which consists of your access and site code (80x) and the directory number of your phone (3001).


Task 8: Configure PSTN Backup for Intercluster Calls In this task, you will update the route lists HQ-ICT-HQ_rl, HQ-ICT-BR_rl, BR-ICT-HQ_rl, and BR-ICT-BR_rl to allow the use of a local gateway in case of failure of the preferred H.225 trunk.

Step 120 Choose Call Routing > Route/Hunt > Route List, click Find. Step 121 Edit the route lists HQ-ICT-HQ_rl and HQ-ICT-BR_rl:

Click Add Route Group and select HQ_rg from the list. Make sure that the route group H225-Trunk_rg is first. Click Save, then OK and reset the lists.

Step 122 On your HQ router configure a translation rule for the called numbers which changes the site code numbers (80y2XXX and 80y3XXX) to the full 10 digit PSTN numbers and a leading PSTN access code 9 and long distance 1 (where y is the neighbor pod number):

voice translation-rule 4 rule 1 /^80y2/ /9151y5552/ rule 2 /^80y3/ /9152y5553/

Configuration 5-9: Translation rule of the called numbers

Note The PSTN access code 9 will be removed at the outgoing dial peer. The calling party number is already configured to be expanded to a full 10 digit PSTN number by the translation profile that is applied at the voice port (pstn-out).

Step 123 Bind the translation rule to the ict-backup profile, configure an incoming dial peer that matches the incoming called numbers starting with 80y (where y is the neighbor pod number), and bind the translation profile to the dial peer.

voice translation-profile ict-backup translate called 4 exit dial-peer voice 4 voip incoming called-number 80y.... voice-class codec 1 translation-profile incoming ict-backup end

Configuration 5-10: Bind the translation rule to the ict backup profile

Step 124 In CUCM Administration, select Call Routing > Route/Hunt > Route List, click Find and select the route list BR-ICT-HQ_rl.

Step 125 Click Add Route Group and select BR_rg from the drop-down list. Step 126 In the Calling Party Transformations field enter 52x5553XXX (where x is your pod number). Step 127 In the Called Party Transformations select NANP:PreDot from the Discard Digits list and

enter 151y555 in the Prefix Digits (Outgoing Calls) field (where y is the neighbor pod number). Step 128 Click Save and then OK. Make sure that the route group H225-Trunk_rg is first and reset the

route list. Step 129 Go to Back to Find/List and select the route list BR-ICT-BR_rl. Step 130 Click Add Route Group and select BR_rg from the drop-down list. Step 131 In the Calling Party Transformations field enter 52x5553XXX (where x is your pod number). Step 132 In the Called Party Transformations select NANP:PreDot from the Discard Digits list and

enter 152y555 in the Prefix Digits (Outgoing Calls), where y is the neighbor pod number. Step 133 Click Save, then OK. Make sure that the route group H225-Trunk_rg is first in the list and reset

the route list.


Verification Step 134 On your HQ router, shutdown the sub-interfaces connecting to the routers GK and other HQ. In

pod1 these are: Serial0/0/0.103 and Serial0/0/0.105. In pod2 these are Serial0/0/0.503 and Ser 0/0/0.501.

Step 135 On your HQ router, enable debug ras and debug isdn q931. Step 136 From Phonex-1 or Phonex-2, call a HQ and a BR phone located in the other pod. Use 80y2001 or

80y2002 to dial a HQ phone and 80y3001 to dial the BR phone. The other phone should ring and the E.164 PSTN number of the calling phone should be shown as the source of the call.

Note Calls to HQ display the 91 prefix at the beginning of the calling number, calls to BR do not.

Step 137 On your HQ gateway, check the debug output to see that the call was using the backup path (i.e. the HQ gateway) and that calling and called party number have been transformed to PSTN numbers.

Step 138 On your BR router, enable debug isdn q931. Step 139 From Phonex-3, call a HQ and a BR phone located in the other pod. Use 80y2001 or 80y2002 to

dial a HQ phone and 80y3001 to dial the BR phone. The other phone should ring and the E.164 PSTN number of the calling phone should be shown as the source of the call.

Note Calls to HQ display the 91 prefix at the beginning of the calling number, calls to BR do not.

Step 140 On your BR gateway, check the debug output to see that the call was using the backup path (i.e. the BR gateway) and that calling and called party number have been transformed to PSTN numbers.

Step 141 On your HQ router reactivate the serial sub-interfaces.

Note You may need to reset the H.225 trunk in CUCM to re-register with the gatekeeper after the breakdown.

Step 142 Place intersite calls again. They should not use the backup paths (PSTN gateways HQ and BR anymore). You can verify this by watching debug isdn q931 output on these two routers.


Task 9: Configure Tail-End Hop Off within the Pod In this task, you will configure a dial plan that allows Tail-End Hop Off (TEHO) for calls to the 51x and 52x PSTN area codes. This means, that BR phones will first try the HQ PSTN gateway when calling PSTN numbers in area 51x and HQ phones will first try the BR PSTN gateway when calling PSTN numbers in area 52x. Both will use the local PSTN gateway as a backup.

Step 143 Choose Call Routing > Route/Hunt > Route List and Add New. Step 144 Create a route list named HQ-TEHO-BR_rl with CUCM Group Default and click Save. Step 145 Click Add Route Group and select BR_rg from the list. Step 146 Enter 52x5553001 for the Calling Party Transform Mask and XXXXXXX for the Called

Party Transform Mask. Save and click OK.

Note BR extension 3001 will act as the attendant for the TEHO return calls.

Step 147 Back in the Route List settings click again Add Route Group, select HQ_rg. Click Save and then OK.

Step 148 Go to Call Routing > Route/Hunt > Route Pattern and click Add New. Step 149 Create a new pattern 9.152x[2-9]XXXXXX (where x is your pod number). Use route partition

HQ_PSTN, description HQ TEHO and point it to Route List HQ-TEHO-BR_rl. Click Save and OK.

Note This route pattern is available to HQ phones in order to prefer the BR gateway over the HQ gateway.

Step 150 Go to Call Routing > Route/Hunt > Route List and click Add New. Step 151 Create a route list named BR-TEHO-HQ_rl with CUCM group Default. Click Save. Step 152 Click Add Route Group and select HQ_rg from the list. Click Save and then OK. Step 153 Click Add Route Group and select BR_rg from the list. Step 154 Enter 52x5553XXX for the Calling Party Transform Mask and select NANP:PreDot for the

Discard Digits in the Called Party Transformations. Click Save and then OK. Step 155 Go to Call Routing > Route/Hunt > Route Pattern and click Add New. Step 156 Create the route pattern 9.151x[2-9]XXXXXX in partition BR_PSTN with description BR

TEHO pointing to route list BR-TEHO-HQ_rl. Click Save and then OK.

Note This route pattern is available to BR phones in order to prefer the HQ gateway over the BR gateway.


Step 157 On your HQ router configure digit manipulation to translate the called long distance number 9-1-51x-XXX-XXXX (where x is your pod number) to a local number: 9-XXX-XXXX. PSTN access code 9 should be preserved.

voice translation-rule 5 rule 1 /^9151x\(.......\)/ /9\1/

Configuration 5-11: On HQrouter configure digit manipulation

Note The PSTN access code 9 will be removed at the existing outgoing dial peer. The calling party number is already configured to be set to the HQ attendant number if the call originates from a different directory number than 2XXX.

Step 158 Bind the newly created rule to a new voice translation profile teho-out, configure an incoming dial peer that matches the incoming called long distance numbers starting with 9151x and bind the translation profile to the dial peer.

voice translation-profile teho-out translate called 5 dial-peer voice 5 voip incoming called-number 9151x....... voice-class codec 1 translation-profile incoming teho-out

Configuration 5-12: Bind the newly created rule to a new voice translation profile

Verification Step 159 On your HQ and BR routers, start isdn 931 debugging. Step 160 Test if TEHO is working from HQ:

From any HQ phone, call any PSTN number in area code 52x (e.g. 9-1-52x-333-4444). The PSTN Phone should ring on the line labeled Local; the incoming calling number should be 52x5553001 should appear on the PSTN phone display.

Check out the debug output on the BR router to make sure that the call went through it. Step 161 Test the backup of TEHO from HQ:

On your BR router, shut down the E1 0/2/0 controller. Try the same call again. This time the call should use the local HQ gateway. The call should

be received at the PSTN phone on the line labeled LD; the calling party number should be the 10 digit number of the calling phone.

Step 162 Reactivate the E1 controller on your BR router. Step 163 Verify if TEHO is working from BR phones:

Place a call from Phonex-3 to any PSTN number in area code 51x (e.g. 9-1-51x-333-4444). The PSTN Phone should ring on the line labeled Local; the incoming calling number should be 51x5552001.

Check out the debug output on the HQ router to make sure that the call went through it. Step 164 Test the backup of TEHO from BR:

On your HQ router, shut down Loopback0. Try the same call again. This time the call should use the BR gateway. The call should be

received at the PSTN phone on the line labeled LD; the calling party number should be the 10 digit number of the calling phone.

On your HQ router, reactivate Loopback0 and turn off debugging on both gateways.


Task 10: Configure SIP Calls to an ITSP In this task, you will configure a dial plan for calls which are prefixed with a carrier selection code indicating that the call should be sent via an ITSP.

Step 165 Select Call Routing > Route/Hunt > Route Group and click Add New. Step 166 Configure a route group named SIP-Trunk_rg and add SIP-Trunk to it. Click Save. Step 167 Choose Call Routing > Route/Hunt > Route List and click Add New. Step 168 Configure a route list named HQ-ITSP_rl with description HQ Access to ITSP and CUCM

group Default. Click Save. Step 169 Click Add Route Group and select SIP-Trunk_rg from the list.

Note When sending calls to the ITSP the prefix 9 should not be sent. The carrier selection information (1010-xxx), however, should be preserved. Therefore discard digits instruction PreDot is used at the SIP_Trunk_rg while discard digits instruction PreAt is used at the HQ_rg (where the carrier selection information has to be removed).

Step 170 Enter 51x5552XXX for the Calling Party Transform Mask and select NANP:PreDot from the Discard Digits list in the Called Party Transformations. Click Save and then OK.

Step 171 Click Add Route Group and select HQ_rg from the list.

Note Digit manipulation must remove the access code 9 and the carrier selection information from the called number to become a standard PSTN number. It must prefix a PSTN access code 9. The calling party number does not need to be modified here as this is already configured at the H.323 gateway. The PSTN access code 9 that is prefixed here will be stripped off at the H.323 gateway.

Step 172 In the Called Party Transformations select NANP:PreAt from the Discard Digits list and enter 9 in the Prefix Digits (Outgoing Calls) field.

Step 173 Click Save and then OK. Step 174 Go to Call Routing > Route/Hunt > Route Pattern and click Add New. Step 175 Create a route pattern 9.1010111@ in partition HQ_SIP with description HQ to ITSP and

numbering plan NANP, pointing to route list HQ-ITSP_rl. Click Save and then OK. Step 176 Choose Call Routing > Route/Hunt > Route List and click Add New. Step 177 Create a route list named BR-ITSP_rl with description BR Access to ITSP and CUCM group

Default. Click Save. Step 178 Click Add Route Group and select SIP-Trunk_rg from the list.

Note Digit manipulation must remove the access code 9 from the called number and change the calling four digit directory number to a 10 digit PSTN number.

Step 179 Enter 52x5553XXX for the Calling Party Transform Mask (where x is your pod number) and select NANP:PreDot from the Discard Digits list in the Called Party Transformations.



Step 181 Click Add Route Group and select BR_rg from the list.

Note Digit manipulation must remove the access code 9 and the carrier selection information from the called number to create a standard PSTN number, and change the calling four digit directory number to a 10 digit PSTN number.

Step 182 In the Calling Party Transformation enter 52x5553XXX for the Calling Party Transform Mask and select NANP:PreAt from the Discard Digits list in the Called Party Transformations.

Step 183 Click Save and then OK. Step 184 Go to Call Routing > Route/Hunt > Route Pattern and click Add New. Step 185 Create the route pattern 9.1010111@ in route partition BR_SIP with description BR to ITSP

and numbering plan NANP pointing to route list BR-ITSP_rl. Click Save and then OK.

Verification Step 186 Verify that your HQ phones can place calls to the ITSP:

From a HQ phone dial a PSTN number prefixed with the SIP provider code (e.g. 9-1010-111-555-1234).

The call should be received at the PSTN phone at the line labeled SIP; the calling party number should be the 10 digit E.164 number of the calling phone.

Step 187 Verify if your BR phones can place calls to the ITSP: From Phonex-3 dial a PSTN number prefixed with the SIP provider code (e.g. 9-1010-111-

555-1234). The call should be received at the PSTN phone at the line labeled SIP; the calling party

number should be the 10 digit E.164 number of the calling phone. Step 188 Verify that your backup through the PSTN is working correctly for HQ and BR phones:

On your HQ router, shutdown the sub-interfaces connecting to the routers GK and other HQ. In pod1 these are: Serial0/0/0.103 and Serial0/0/0.105. In pod2 these are Serial0/0/0.503 and Serial 0/0/0.501.

Enable isdn q931 debugging on your HQ and BR routers. Repeat a call from a HQ phone to the ITSP. The call should be received at the PSTN phone on a different line than the one labeled SIP.

The actual line depends on the type of number you dialed after the carrier selection (local, long distance, etc., in case of the number 9-1010-111-555-1234, the local line will ring. The calling party number should be the 10 digit E.164 number of the calling phone.

Check the debug output at the HQ gateway that the PSTN was used for the call as the SIP connection is broken.

Place the same call from Phonex-3, and watch the debug output on your BR gateway. The call should go through the BR gateway.

Reactivate the serial subinterfaces on the HQ router and disable all debugging.


Task 11: Implement a Cisco Unified Border Element for Calls to the ITSP

In this task, you will change the SIP trunk configuration to point to your HQ router instead of directly pointing to the ITSP. The Cisco Unified Border Element will be configured at HQ in order to act as a signaling and media proxy towards the ITSP. This way, the IP addresses of Cisco Unified Communications Manager and all IP phones can be hidden from the outside. Only the Cisco Unified Border Element needs to have connectivity to the ITSP.

Step 189 Go to Device > Trunk and click Find. Step 190 Select SIP-Trunk and change the Destination Address from 192.168.3.1 to 192.168.x.1 (the

loopback of your HQ router). Step 191 Click Save and reset the trunk. Step 192 On your HQ router, enable SIP to SIP connections and bind the SIP protocol to the loopback

interface. voice service voip allow-connections sip to sip sip bind control source-interface Loopback0 bind media source-interface Loopback0

Configuration 5-13: On HQrouter enable SIP

Step 193 On your HQ router, configure appropriate incoming and outgoing SIP dial peers: dial-peer voice 6 voip session protocol sipv2 incoming called-number 1010111T ! dial-peer voice 7 voip destination-pattern 1010111T session protocol sipv2 session target ipv4:192.168.3.1

Configuration 5-14: ON HQrouter configure appropriate incoming

Verification Step 194 On your HQ router enable debug ccsip messages. Step 195 Place calls to the ITSP (e.g. 9-1010-111-555-1234) and watch the debug output at HQ. You

should see messages being exchanged between CUCM and the Cisco Unified Border Element as well as between the Cisco Unified Border Element and the ITSP router (192.168.3.1).

Step 196 While having an active call, enter show voip rtp connections on your HQ router. You should see two voip call legs: one between CUCM and Cisco Unified Border Element and one between the Cisco Unified Border Element and the ITSP router.

Step 197 Turn off all debugging.


Task 12 (Optional): Configure Tail-End Hop-Off between Pods In this task, you will configure a dial plan that allows tail-end hop-off (TEHO) for calls to the 51y and 52y PSTN area codes adjacent to the gateways of the other pod. This means, that your HQ and BR phones will first try the H.225 trunk when calling PSTN numbers in area 51y or 52y. Both will use their local PSTN gateway as a backup.

Note The y in the area codes is the number of the neighbour pod and x is your pod.

Step 198 Go to Device > Trunk, click Find and select H225-Trunk-x. Step 199 In the Inbound Calls pane change the Significant Digits from 4 to All. Click Save and reset the

trunk.

Note Until now calls through the intercluster trunk were addressed to internal phones only. The called number was 10 digits long. At the trunk, significant digits were set to 4 in order to get from the E.164 10 digit number to 4 digit directory numbers. This is changed to allow TEHO calls (destined for PSTN area codes 51x and 52x) to be recognized. The translation patterns will be used for incoming calls; they replace the significant digits 4 setting at the trunk.

Step 200 Go to Call Routing > Translation Pattern and click Add New. Step 201 Create the translation pattern 51x5552XXX in partition ICT_IN with description Incoming ICT

Calls to HQ Phones. Set the CSS to Inbound-ICT_css and the Called Party Transform Mask to 2XXX.

Step 202 Click Save and then Copy. Modify these parameters, and then save: Translation Pattern: 52x5553XXX Description: Incoming ICT Calls to BR Phones Called Party Transform Mask: 3XXX

Note At this point, verify that intercluster calls are working with the modified configuration (translation patterns for calls received through the trunk instead of significant digits set to 4): Place a call from a HQ phone of your pod to 80y2001 and 80y3001. These calls should work and the calling number should be 80x followed by the directory number of the phone you used for placing the call.

Step 203 Go to Call Routing > Route/Hunt > Route List and click Add New.

Note Next you will create a new route list to allow HQ phones to use TEHO to PSTN destinations close to the gateways of the other pod (area codes 51y and 52y). The calling number does not need to be modified as it will be replaced by the number of an attendant located at the site where the call breaks out to the PSTN. The called number is modified to a 10 digit number as the gatekeeper routes calls based on code prefixes. If the call does not succeed over the intercluster trunk, the local gateway (HQ) will be used. Digit manipulation for the backup path is already in place.

Step 204 Create a route list named HQ-TEHO-OtherPod_rl with description HQ TEHO Access to other pod, and CUCM group Default. Click Save.

Step 205 Click Add Route Group and select H225-Trunk_rg from the list.


Step 206 Enter XXXXXXXXXX (10 digits) in the Called Party Transform Mask. Click Save and then OK.

Step 207 Back in the Route List Configuration window click Add Route Group again and select HQ_rg from the list. Click Save and then OK.

Step 208 Go to Call Routing > Route/Hunt > Route Pattern and click Add New. Step 209 Create the route pattern 9.15[12]y[2-9]XXXXXX in partition HQ_ICT with description HQ

TEHO to other pod pointing to route list HQ-TEHO-OtherPod_rl. Click Save and then OK.

Note Next you must configure a route list that allows BR phones to use TEHO to PSTN destinations close to the gateways of the other pod (area codes 51y and 52y). The calling number does not need to be modified as it will be replaced by the number of an attendant located at the site where the call breaks out to the PSTN. The called number is modified to a 10 digit number as the gatekeeper routes calls based on are code prefixes. If the call does not succeed over the intercluster trunk, the local gateway (BR) is used. In this case the calling party number is changed to a 10 digit number and the PSTN access code 9 is removed from the called number.

Step 210 Go to Call Routing > Route/Hunt > Route List and click Add New. Step 211 Create a route list named BR-TEHO-OtherPod_rl, with description BR-TEHO Access to

other pod, and CUCM group Default. Click Save. Step 212 Click Add Route Group and select H225-Trunk_rg from the list. Step 213 Enter XXXXXXXXXX (10 digits) for the Called Party Transform Mask. Click Save and then

OK. Step 214 Back in the Route List Configuration window click Add Route Group again, and select BR_rg

from the list. Step 215 Enter 52x555 in the Prefix Digits (Outgoing Calls), of the Calling Party Transformations,

1XXXXXXXXXX (digit 1 and 10 digits) for the Called Party Transform Mask and select NANP:PreDot for the Discard Digits in the Called Party Transformations pane. Click Save and then OK.

Step 216 Go to Call Routing > Route/Hunt > Route Pattern and click Add New. Step 217 Create the route pattern 9.15[12]y[2-9]XXXXXX in partition BR_ICT with description BR-

TEHO to other pod pointing to route list BR-TEHO-OtherPod_rl. Click Save and then twice OK.

Note Next you will configure a route list that allows TEHO calls destined to PSTN numbers in area 51x to be sent out through the HQ gateway if received through the H.225 trunk and initiated in the other pod. The called party number should be changed to a 7 digit number and the calling party number should be set to 51x5552001.

Step 218 Go to Call Routing > Route/Hunt > Route List and click Add New. Step 219 Create a route list named OtherPod-TEHO-HQ_rl with description Incoming TEHO Access

to HQ-near-PSTN from OtherPod and CUCM group Default. Click Save. Step 220 Click Add Route Group and select HQ_rg from the list. Step 221 Enter 51x5552001 for the Calling Party Transform Mask, XXXXXXX (7 digits) for the

Called Party Transform Mask and 9 in the Prefix Digits (Outgoing Calls) field. Click Save and then OK.

Step 222 Go to Call Routing > Route/Hunt > Route Pattern and click Add New.


Step 223 Create the route pattern 51x[2-9]XXXXXX in partition ICT_IN, with description Incoming TEHO-Calls from other Pod to HQ-near-PSTN pointing to route list OtherPod-TEHO-HQ_rl.


Note Next you will configure a route list that allows TEHO calls destined to PSTN numbers in area 52x to be sent out through the BR gateway if received through the H.225 trunk and initiated in the other pod. The called party number should be changed to a 7 digit number and the calling party number set to 52x5553001.

Step 225 Go to Call Routing > Route/Hunt > Route List and click Add New. Step 226 Create a route list named OtherPod-TEHO-BR_rl with description Incoming TEHO Access to

BR-near-PSTN from OtherPod and CUCM group Default. Click Save. Step 227 Click Add Route Group and select BR_rg from the list. Step 228 Enter 52x5553001 for the Calling Party Transform Mask and XXXXXXX (7 digits) for the

Called Party Transform Mask. Click Save and then OK. Step 229 Go to Call Routing > Route/Hunt > Route Pattern and click Add New. Step 230 Create the route pattern 52x[2-9]XXXXXX in partition ICT_IN with description Incoming

TEHO Calls from other Pod to BR-near-PSTN pointing to route list OtherPod-TEHO-BR_rl. Click Save and then OK.

Verification

Note Perform this verification together with the students from the other pod.

Step 231 Activate isdn q931 debugging at your HQ and BR gateways to verify that the call was sent over the intercluster trunk. On the GK router use debug ras to see that the call is routed via the gatekeeper.

Step 232 From a HQ phone, call a PSTN number adjacent to the HQ router of the other pod (e.g. 9-1-51y-333-4444). The PSTN Phone in the other pod should receive the call at the line labeled Local; the calling party number should be 51y5552001.

Step 233 Place additional test calls, this time to a PSTN number adjacent to the BR router of the other pod (e.g. 9-1-52y-333-4444). The call should again be received at the line labeled Local; this time the calling party number should be 52y5553001.

Step 234 Repeat the above tests from Phonex-3. You should get identical results. Step 235 Verify the backup option - that calls to PSTN destinations in area codes 51y and 52y use the

local PSTN gateway when the H.225 trunk fails: On your HQ router, shutdown the sub-interfaces connecting to the routers GK and other HQ.

In pod1 these are: Serial0/0/0.103 and Serial0/0/0.105. In pod2 these are Serial0/0/0.503 and Serial 0/0/0.501.

Repeat the test calls and verify that the calls are routed out at the local gateway (i.e. HQ if the call was placed from a HQ phone and BR if the call was placed from a BR phone). The call should be received at your PSTN phone at the line labeled LD; the calling party number should be the 10 digit E.164 number of the calling phone.


Appendix: Launching CIPC-emulated Phones with Sound

Note The CIPC-emulated phones start automatically with the option “Sound on remote computer”. To hear sounds, you must create and use an RDP session with option “Sound on the local computer”.

Step 236 In your E-learning desk, click Edit Profile and in the Terminal Service Client (RDP) options, check Use Native OS client and save profile.

Figure 5-1: RDP native OS client settings

Step 237 Click on the Phone icon in the topology picture. When you are asked if you want to Save, Run or Cancel the program, select Save. Change name from RDPconnection.rdp to a meaningful name for that phone.


Step 238 Right-click the RDP connection and select Edit. In the General tab, enter the username and password: administrator/admin, check the Save my password checkbox.

Figure 5-2: Expanded RDP Connection window


Step 239 Select Local Resources tab and set the Remote Computer sound option to Bring to this computer.

Figure 5-3: Bring to this computer sound option

Step 240 Go back to the General tab and re-save the connection. Step 241 Launch the RDP connection by double-clicking it.


CIPT260

Implementing SRST and MGCP Fallback

1. Objective In this exercise you will configure SRST to provide call survivability for Cisco IP phones, and MGCP fallback for gateway survivability. To achieve this goal, you will add an SRST reference, configure a device pool with the SRST reference and apply the device pool to remote phones. Then you will configure SRST and MGCP Fallback support at the Cisco IOS Gateway.

You will configure Call Forward UnRegistered (CFUR) for remote phones to allow phones located in the main site to call remote phones via the PSTN during SRST mode, by creating a CFUR CSS, adjusting CFUR parameters and updating the phones.

You will configure a dial plan at the SRST gateway which allows incoming calls placed to the E.164 PSTN number of the remote phones to be sent to the appropriate directory number. Finally you will implement PSTN access for remote users and allow them to place calls to headquarter phones over the PSTN by dialing internal directory numbers.

































Parameter Value



10.x.5.0/24


10.x.4.0/24




PSTN LAN 10.3.0.0/24





4. Detailed Instructions Follow the steps in the tasks to implement SRST and MGCP Fallback.

Task 1: Configuring SRST Gateways in Cisco Unified Communications Manager

In this task you will add an SRST references, configure a device pool with the SRST reference and apply the device pool to remote phones.

Step 1 Connect to your Phonex-1 (x is your pod number) and login in as administrator with password admin. Access CUCM Administration by clicking its desktop shortcut (https://10.x.1.1/ccmservice). Log in as CCMAdministrator with password appuserpass. Update the MAC addresses of your classroom phones as needed.


Step 2 Go to System > SRST and click Add New to create a new SRST reference with these settings (and save it):

Name: BR_SRST Port: 2000 (default) IP Address: 192.168.x.2 (loopback0 of the BR router)

Step 3 Navigate to System > Device Pool and click Add New to create a new device pool with these parameters (and save it):

Device Pool Name: Branch Cisco Unified Communications Manager Group: Default Date/Time Group: CMLocal Region: Default SRST Reference: BR_SRST

Step 4 Go to Device > Phone, click Find, and select Phonex-3. Step 5 Set the Device Pool to Branch. Click Save, then OK and reset the phone.

Verification Step 6 On Phonex-3 press the Settings button, and select Network (Device) Configuration and scroll

down to the CallManager section. Step 7 The second entry should say CallManager 2 SRST and show the loopback IP address of your BR

router (192.168.x.2).


Task 2: Configure a Cisco IOS Gateway for MGCP Fallback and SRST

In this task, you will configure SRST and MGCP Fallback support at a Cisco IOS Gateway.

Step 8 On your BR router configure the SRST feature: call-manager-fallback ip source-address 192.168.x.2 max-dn 2 dual-line max-ephones 2

Configuration 4-1: Configure the SRST feature

Step 9 On the BR router enable the gateway fallback feature:

ccm-manager fallback-mgcp

Configuration 4-2: Enable gateway fallback feature

Step 10 Configure the default voice application (H.323) to take over if the MGCP application is not available:

application global service alternate Default

Configuration 4-3: Configuere the default voice application

Verification Step 11 On your BR router start ephone registration debugging using the command debug ephone

register. Step 12 On BR router, shut down the WAN sub-interface (in pod1: Serial0/0/0.401, in pod2:

Serial0/0/0.805). Phonex-3 should register with the SRST router - the phone will show ‘CM Fallback Service Operating’ at the bottom of the display.

Step 13 On the BR gateway view the debug output indicating that the phone registered with the SRST gateway.

Task 3: Implement a Dial Plan in Cisco Unified Communications Manager Supporting Outbound Calls during SRST Mode

In this task, you will configure CFUR for remote phones to allow phones located in the main site to call remote phones via the PSTN during SRST mode.

Step 14 Navigate to Call Routing > Class of Control > Calling Search Space and click Add New. Step 15 Create a new CSS named CFUR_css and description ‘CFUR for BR Phones’. Add the partition

HQ_PSTN and click Save. Step 16 Go to System > Service Parameters, select your CUCM (10.x.1.1) and choose the Cisco

CallManager service. Step 17 In the Clusterwide Parameters (Feature - Forward) pane change the Max Forward

UnRegistered Hops to DN parameter to 2 (default is 0) and click Save. Step 18 Go to Call Routing > Directory Number, click Find, and select directory number 3001 which

is in the BR_Phones partition.


Step 19 Scroll to the Call Forward and Call Pickup Settings pane and enter the following parameters in the Forward Unregistered Internal and in the Forward Unregistered External rows:

Destination: 9152x5553001 Calling Search Space: CFUR_css

Step 20 Save the directory number.

Verification Step 21 On your BR router enable isdn debugging. Step 22 From one of your HQ phones call 3001. The call will not be extended to Phonex-3. On the

calling phone, enter manually 3001, Phonex-3 will ring.

Note If you use CIPC-emulated phones instead of classroom IP phones and want to hear sounds, follow the procedure in the appendix.

Step 23 Verify in the debug output if the call hits the BR gateway. The call arrives on the BR router but direct-inward-dial is not enabled and the called party number is a 10 digit number and not a 4 digit directory number.

Note BR gateway accepts the call and as direct-inward-dial is not enabled it waits for dialed digits (two stage dialing).

Task 4: Implement a Dial Plan at the SRST Gateway Supporting Inbound and Outbound Calls when being in MGCP Fallback and/or in SRST Mode

In this task, you will configure a dial plan at the SRST gateway which allows incoming calls placed to the E.164 PSTN number of the BR phones to be sent to the appropriate directory number. In addition you will implement PSTN access for BR users. Finally you will allow BR users to place calls to HQ phones over the PSTN by dialing internal directory numbers.

Note In the presented configurations x represents your pod number and y represents the number of the neighbor pod.

Step 24 On your BR router enable direct inward dial on the PSTN voice port: dial-peer voice 1 pots incoming called-number 52x5553... direct-inward-dial port 0/2/0:15

Configuration 4-4: Enable direct inwar dial


Step 25 On BR router configure a translation profile to manipulate the incoming called number from the PSTN (the full PSTN number 52x 5553001) to the 4 digit directory number, and bind the newly created translation profile to the PSTN voice port:

voice translation-rule 1 rule 1 /^52x5553/ /3/ exit voice translation-profile pstn-in translate called 1 exit voice-port 0/2/0:15 translation-profile incoming pstn-in

Configuration 4-5: ON BR router configure translation profile

Step 26 Verify that HQ phones can call 3001. Note that the calling party number displayed at Phonex-3 is the PSTN number of the calling phone.

Step 27 On your BR router add a destination pattern to the existing pots dial peer 1 to allow outgoing PSTN calls.

dial-peer voice 1 destination-pattern 9T

Configuration 4-6: On BR router add destination pattern

Step 28 From Phonex-3 call a HQ phone (9-1-511-555-2001 or 9-1-511-555-2002). The call should work.

Step 29 From Phonex-3 call any PSTN number (e.g. 9-1-333-333-4444). Note that the calling party number displayed at the PSTN phone is the 4 digit internal directory number of Phonex-3 (3001).

Step 30 On your BR router configure digit manipulation that adds 52x555 to the 4 digit directory number starting with 3 to ensure that the calling party number on outgoing calls uses E.164 PSTN format:

voice translation-rule 2 rule 1 /^3/ /52x5553/ exit voice translation-profile pstn-out translate calling 2 exit voice-port 0/2/0:15 translation-profile outgoing pstn-out

Configuration 4-7: On BR router configure digit manipulation

Step 31 Verify that outgoing calls have an E.164 calling number (52x-555-3001).

Note The translation between internal directory numbers and external PSTN numbers could be replaced by the dialplan-pattern 1 52x5553…extension-length 4 command, entered under call-manager-fallback.

Step 32 On the BR router, configure an outgoing dial peer that matches the internal directory number of HQ phones and add the appropriate prefix. This will allow calls to other sites using internal numbers instead of PSTN numbers.

dial-peer voice 2000 pots destination-pattern 2... port 0/2/0:15 forward-digits all prefix 151x555

Configuration 4-8: On BR router configure an outgoing dial peer

Step 33 From Phonex-3 call a HQ phone using its 4 digit internal directory number.


Step 34 Configure an outgoing dial peer that matches the HQ of the other pod and modify the called number appropriately:

dial-peer voice 8022 pots destination-pattern 80y2... port 0/2/0:15 forward-digits 4 prefix 151y555

Configuration 4-9: Configure outgoing dial peer that mach the HQ

Step 35 From Phonex-3 call a HQ phone in the other pod using site code dialing (80y-2001 or 80y-2002). Step 36 On the BR router configure a dial peer for the other pod BR and modify the called number

appropriately: dial-peer voice 8023 pots destination-pattern 80y3... port 0/2/0:15 forward-digits 4 prefix 152y555

Configuration 4-10: Configure dial peer

Step 37 From Phonex-3 call the BR phone of the other pod (80y-3001).













Note If you cannot hear audio when the phone rings, reset CIPC in the session with audio left on the remote computer.



CIPT260

Implementing Cisco Unified Communicma boliations Manager Express as SRST Fallback

1. Objective In this exercise you will configure Cisco Unified Communications Manager Express in SRST mode to provide basic telephony services to phones that lost the connection to Cisco Unified Communications Manager. You will change from standard SRST to Cisco Unified Communications Manager Express in SRST fallback mode. Finally you will configure Cisco Unified Communications Manager Express to provide MOH to Cisco IP phones.

































Parameter Value



10.x.5.0/24


10.x.4.0/24




PSTN LAN 10.3.0.0/24





4. Detailed Instructions Follow the steps in the tasks to implement Cisco Unified Communications Manager Express as SRST Fallback.

Task 1: Configure Cisco Unified Communications Manager Express in SRST Fallback Mode

In this task you will change from standard SRST to Cisco Unified Communications Manager Express in SRST fallback mode.

Step 1 Connect to your Phonex-1 (x is your pod number) and login in as administrator with password admin. Access CUCM Administration by clicking its desktop shortcut (https://10.x.1.1/ccmadmin). Log in as CCMAdministrator with password appuserpass. Update the MAC addresses of your classroom phones, if needed.


Step 2 On your BR router, delete the SRST settings: no call-manager-fallback

Configuration 4-1: On BRrouter delete SRST settings

Note The dial peers and translation profiles configured in the standard SRST lab will be reused for UCME in SRST Fallback mode.

Step 3 On the BR router, enable Cisco Unified Communications Manager Express in SRST mode: telephony-service ip source-address 192.168.x.2 system message CUCME in SRST Mode max-ephones 2 max-dn 2 srst mode auto-provision all ! Command above saves the learned ephone’s and ephone-dn’s srst dn line-mode dual srst ephone description SRST learned create cnf-files ! Command above makes the router generate configuration files

Configuration 4-2: On BR router enable CUCME in SRST mode


Verification Step 4 On your BR router, verify the SRST fallback configuration using the show telephony-service

command. Step 5 Verify the current files accessible to IP phones using the show telephony-service tftp-bindings

command. Step 6 On BR router, break connectivity to CUCM by shutting down the WAN interface (in pod1:

Serial0/0/0.401, in pod2: Serial0/0/0.805). Step 7 Wait about a minute. From a HQ phone call Phonex-3 (3001). The call should work; the calling

party number should be the 10 digit PSTN number.

Note If you use CIPC-emulated phones instead of classroom IP phones and want to hear sounds, follow the procedure in the appendix.

Step 8 From Phonex-3 call a HQ phone (use internal dialing: 2001 or 2002). The call should work; the calling party number should be the 10 digit PSTN number of Phonex-3.

Step 9 On your BR router, verify the ephone-dn and ephone in the configuration using the show running-config command.

Note Next time the phone registers with Cisco Unified Communications Manager Express, Cisco Unified Communications Manager Express uses the stored configuration instead of learning the phone configuration using SNAP. In order to configure a phone with features that cannot be learned by SRST, you can preconfigure the ephone-dn only (then the ephone is learned) or ephone-dn and ephone.

Task 2: Configure MOH on Cisco Unified Communications Manager Express

In this task you will copy a MoH audio file to the SRST router flash and configure Cisco Unified Communications Manager Express to provide MOH to the phones.

Step 10 On Phonex-3, navigate to C:\ Program Files\tftpsrv and start TFTPSRV.EXE. Set the root drive to D:\8k16bitmono. It contains the file music-on-hold.au

Step 11 On your BR router, download music-on-hold.au using the copy tftp flash command without erasing the flash. TFTP server address is 10.x.6.1.

Step 12 On your BR router, enable MOH: telephony-service moh music-on-hold.au

Configuration 4-3: Enable MOH


Verification Step 13 Verify that the branch phones can listen to MOH when put on hold:

Place a call between a HQ phone and the BR phone (Phonex-3). On Phonex-3 put the call on hold. The HQ phone should play MOH coming from the BR router.
















CIPT260

Implementing Bandwidth Management

1. Objective In this exercise you will configure multicast MOH, regions, local conference bridges, and transcoders to reduce bandwidth requirements on the IP WAN. To achieve this goal, you will first enable the Cisco IP Voice Media Streaming App feature service which provides several software media resources running on Cisco Unified Communications Manager. You will change the default names and descriptions of these media resources. Next you will configure regions in order to prevent audio streams that are sent over the IP WAN from using high-bandwidth codecs. Only G.729 is permitted for streams that traverse the IP WAN.

You will implement a transcoder to allow remote users to join conferences on a G.711-only software conference bridge although the remote phones are not allowed to use G.711 over the IP WAN. The transcoder will transcode the G.729 audio stream received from remote phones to a G.711 stream towards the software conference bridge and vice versa.

You will configure a local hardware conference bridge at the branch. You will implement media resource groups and media resource group lists in order to ensure that IP phones use the local conference media resource.

































Parameter Value



10.x.5.0/24


10.x.4.0/24




PSTN LAN 10.3.0.0/24





4. Detailed Instructions Follow the steps in the tasks to implement bandwidth management.

Task 1: Enable Software Media Resources on Cisco Unified Communications Manager

In this task you will enable the Cisco IP Voice Media Streaming App feature service which provides several software media resources running on CUCM. You will change the default names and descriptions of these media resources.

Step 1 Connect to your Phonex-1 (x is your pod number) and login in as administrator with password admin. Access CUCM Serviceability by clicking its desktop shortcut (https://10.x.1.1/ccmservice). Log in as CCMAdministrator with password appuserpass. Update the MAC addresses of your classroom IP phones as needed.


Step 2 Go to Tools > Service Activation and activate the Cisco IP Voice Media Streaming App service.

Note This service provides the following software media resources: Annunciator, Conference Bridge, Media Termination Point, and Music on Hold Server.

Step 3 Go to CUCM Administration and update the MAC addresses of the classroom phones, if needed. Step 4 Choose Media Resources > Annunciator, click Find. Step 5 Click the available annunciator (ANN_2). Change its name to HQ-SW-ANN and the description

to ‘Software ANN’. Click Save. Step 6 Choose Media Resources > Conference Bridge, click Find, select CFB_2. Rename it to HQ-

SW-CFB with description ‘Software CFB’ and save. Step 7 Choose Media Resources > Media Termination Point, click Find, select MTP_2. Rename it

to HQ-SW-MTP with description ‘Software MTP’ and save. Step 8 Choose Media Resources > Music On Hold Server, click Find, select MOH_2. Rename it to

HQ-SW-MOH with description ‘Software MOH’ and save.

Verification

Note If you want to hear sounds when placing calls to CIPC-emulated phones, follow the procedure in the appendix.

Step 9 Create an ad-hoc conference with Phonex-1, Phonex-2, and Phonex-3 as members. (From Phonex-1 establish a call to Phonex-2, press Confrn softkey, establish a call to Phonex-3 and press Confrn softkey again). This conference uses the HQ-SW-CFB conference resource. Drop the conference call.

Step 10 Make a call between two phones and put the call on hold. The user put on hold hears music.


Task 2: Configure Regions In this task you will configure regions in order to prevent audio streams that are sent over the IP WAN from using high-bandwidth codecs. Only G.729 will be permitted for streams that traverse the IP WAN. The regions will be applied to devices using device pools.

Step 11 In CUCM Administration, go to System > Region and click Find. Step 12 Select region Default. Change its name to HQ, verify that G.711 Audio Codec is allowed for

calls within region HQ and click Save. Step 13 Click Add New to create a new region named Trunks and click Save. Step 14 Using the Modify Relationship to other Regions pane allow G.729 for calls within region

Trunks by highlighting region Trunks in the Regions list and selecting G.729. Click Save. Step 15 Using the same technique, allow G.729 for calls between regions Trunks and HQ.

Note You have to click Save after each change in the Modify Relationship to other Regions pane. The changes will then appear in the Region Relationships pane.

Step 16 Create a new region named BR and save it. Step 17 Allow G.711 for calls within region BR, G.729 for calls between regions BR and Trunks and

also G.729 for calls between regions BR and HQ. Step 18 Go to System > Device Pool, click Find and select the Default device pool and verify that the

region is set to HQ. Step 19 At the related links select Back To Find/List and click Go. Step 20 Select the Branch device pool. Change its region to BR, save and reset. Step 21 Create a new device pool named Trunks, with CUCM group Default, Date/Time Group:

CMLocal, region: Trunks, and SRST Reference: Disable. Click Save. Step 22 Go to Device > Phone and click Find. Verify that Phonex-1 and Phonex-2 are listed with device

pool Default and Phonex-3 is listed with device pool Branch.

Note Regions cannot be directly applied to devices. You have to create different device pools with these regions and then apply the appropriate device pools to the devices.

Step 23 Go to Device > Trunk and click Find. Select the SIP-Trunk. Step 24 Change the device pool from Default to Trunks, click Save and reset the trunk. Step 25 Using the same procedure, configure H225-Trunk-x to use device pool Trunks. Step 26 Go to Device > Gateway, and click Find. Step 27 Verify that the HQ gateway (192.168.x.1) is listed with device pool Default. Step 28 Select the MGCP endpoint of gateway BRx. Change its device pool from Default to Branch.

Save and reset the gateway.


Step 29 On your HQ router (H.323 gateway), configure the inbound dial-peer from the CUCM to prefer G.711 codecs before G.729.

voice class codec 1 codec preference 1 g711ulaw codec preference 2 g711alaw codec preference 3 g729r8 ! dial-peer voice 1 voip destination-pattern 2... voice-class codec 1 exit

Configuration 4-1: On HQ router configure the inbound dial peer

Verification Step 30 Place test calls between the following phones and while being in a call press the ? button on the

IP phone two times. The IP phone will display call information that includes the codec which is used for the call:

Phonex-1 or Phonex-2 and the ITSP (e.g. 9-1010-111-555-1234): This call should use G.729.

Note You can also view information about active calls of an IP phone by use a web browser to browse to the IP address of the IP phone. The built-in web server of the phone provides such (and other) information.

Phonex-1 or Phonex-2 and the PSTN (e.g. 9-911): This call should use G.711. Phonex-1 or Phonex-2 and any phone located in the other pod (e.g. 802-2001): This call

should use G.729. Phonex-1 or Phonex-2: This call should use G.711. Phonex-1 or Phonex-2 and Phonex-3: This call should use G.729. Phonex-3 and the ITSP (e.g. 9-1010-111-555-1234): This call should use G.729. Phonex-3 and any phone located in the other pod (e.g. 802-2001): This call should use

G.729.

Note The specified codecs apply to the primary paths. Other codecs may be used if an alternative path is selected.

Step 31 On Phonex-1 establish an ad-hoc conference with Phonex-2. Try to add Phonex-3 to a conference. This should not work anymore. The only available conference bridge (HQ-SW-CFB) is a software conference and it supports G.711 only. As Phonex-3 is in region BR and this region is not permitted to use G.711 to region HQ (where the software conference bridge is), Phonex-3 cannot join conferences.


Task 3: Implement Transcoders In this task you will implement a transcoder at the HQ in order to allow BR users to join conferences on a G.711-only software conference bridge although BR phones are not allowed to use G.711 over the IP WAN. The transcoder will transcode the G.729 audio stream received from BR phones to a G.711 stream towards the software conference bridge and vice versa.

Step 32 On your HQ router, configure the DSP resources to be used by the CUCM for transcoding: voice-card 0 dspfarm dsp services dspfarm ! sccp local loopback0 sccp ccm 10.x.1.1 identifier 1 version 5.0.1 sccp ! sccp ccm group 1 associate ccm 1 priority 1 associate profile 1 register HQ-HW-XCD ! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 maximum sessions 2 associate application SCCP no shutdown

Configuration 4-2: On HQ router configure DSP resources

Note Next, you will add the transcoder to the CUCM and assign it to a device pool that uses region HQ.

Step 33 In CUCM administration, go Media Resources > Transcoder and click Add New to create a transcoder with these parameters:

Type: Cisco IOS Enhanced Media Termination Point Description: Hardware Transcoder at HQ Name: HQ-HW-XCD Device pool: Default

Note The media resource name must match the router configuration (shown in bold above) and is case sensitive.

Step 34 Click Save.

Verification Step 35 In CUCM administration, verify the registration status of the transcoder HQ-HW-XCD. It

should say ‘Registered with Cisco Unified Communications Manager 10.x.1.1’. Step 36 On your HQ router, check the sccp status with the show sccp command. Verify that the

Transcoding Oper State is ACTIVE and that the TCP Link Status is CONNECTED. Step 37 On your HQ router, check the dspfarm profile status with the show dspfarm profile command.

Verify the status, number of available resources, and the list of supported codecs.


Step 38 Set up an ad-hoc conference with Phonex-1, Phonex-2, and Phonex-3 as members. At each phone press the ? button twice. Phonex-1 and Phonex-2 should show the G.711 codec being used while Phonex-3 shows G.729.

Step 39 On your HQ router check the dsp status with the show dspfarm dsp all command. You should see two used connections representing the two call legs of the transcoder (G.711 to the software conference bridge and G.729 to Phonex-3).

Task 4: Implement a Hardware Conference In this task you will configure a local hardware conference bridge at the branch. You will implement media resource groups and media resource group lists in order to ensure that IP phones use the local conference media resource.

Step 40 On your BR router, configure router DSP resources to be used as hardware Conference Bridge: voice-card 0 dspfarm dsp services dspfarm ! sccp local loopback0 sccp ccm 10.x.1.1 identifier 1 version 5.0.1 sccp ! sccp ccm group 1 associate ccm 1 priority 1 associate profile 1 register BR-HW-CFB ! dspfarm profile 1 conference codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 maximum sessions 1 associate application SCCP no shutdown

Configuration 4-3: ON BR router configure DSP

Note Next, you will add the Cisco IOS Hardware Conference Bridge to CUCM

Step 41 In CUCM administration, 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: BR-HW-CFB Device pool: Branch Device security mode: Non Secure Conference Bridge

Note The media resource name must match the router configuration (shown in bold above) and is case sensitive.

Step 42 Click Save.


Verification Step 43 Verify the registration status of the CFB. It should say ‘Registered with Cisco Unified

Communications Manager 10.x.1.1’. Step 44 On your BR router, check the SCCP status with the show sccp command. Verify that the

Transcoding Oper State is ACTIVE and that the TCP Link Status is CONNECTED. Step 45 On your HQ router, check the dspfarm profile status with the show dspfarm profile command.

Verify the status, number of available resources, and the list of supported codecs.

Task 5: Configure Media Resource Management on CUCM Step 46 In CUCM administration, go to Media Resources > Media Resource Group and click Add

New. Step 47 Create a MRG named HQ-SW-CFB_mrg, and add to it the media resource HQ-SW-CFB

(move to the Selected Media Resources list). Step 48 Create a MRG named BR-HW-CFB_mrg and add to it the media resource BR-HW-CFB. Step 49 Create a MRG named General_mrg and add to it the media resources HQ-HW-XCD, HQ-SW-

ANN, HQ-SW-MOH, HQ-SW-MTP. Step 50 Go to Media Resources > Media Resource Group List and click Add New. Step 51 Create a new MRGL named HQ_mrgl and add to it the MRGs HQ-SW-CFB and General_mrg

(move to the Selected Media Resource Groups list). Step 52 Create a new MRGL named BR_mrgl and add to it the MRGs BR-HW-CFB and General_mrg Step 53 Go to System > Device Pool, click Find, and select device pool Default. Step 54 Select HQ_mrgl as the Media Resource Group List, save and reset the device pool. Step 55 Select Back to Find/List, click Go, and select device pool Trunks. Step 56 Select HQ_mrgl as the Media Resource Group List, save and reset the device pool. Step 57 Select Back to Find/List, click Go, and select device pool Branch. Step 58 Select BR_mrgl as the Media Resource Group List, save and reset the device pool.

Verification Step 59 Initiate an ad-hoc conference from Phonex-3 (adding Phonex-1 and Phonex-2). At each IP phone

press the ? button twice. Phonex-1 and Phonex-2 should use the G.729 codec and Phonex-3 should use the G.711 codec. This is because the conference bridge is at the branch and G.711 is only allowed locally at the branch but not between branch and headquarters. Keep the call open.

Step 60 On your BR router enter the show dspfarm dsp all command. You should see three used connections representing the three conference participants.

Step 61 End the conference. Verify that all DSP resources are freed by entering the command show dspfarm dsp all again.

Step 62 Repeat the previous steps but initiate the conference from Phonex-1 or Phonex-2. This time Phonex-1 and Phonex-2 should use G.711, Phonex-3 should use G.729. The show dspfarm dsp all command will indicate that no conference resources are used at BR. Issue the same command on your HQ router and you will see a transcoder session for the connection of Phonex-3 to the conference bridge.


Task 5: Implement Multicast MOH In this task you will first implement multicast MOH. Then you will implement multicast MOH from branch router flash. This allows BR users to listen to MOH but prevents the MOH streams from being sent over the IP WAN.

Step 63 In CUCM Administration go to Media Resources > Music On Hold Audio Source and click the Find button.

Step 64 Select the only available audio source and verify that the Play Continously checkbox is activated. Activate the Allow Multicasting check box and click Save.

Step 65 Go to Media Resources > Music On Hold Server and click the Find button. Step 66 Click the only available MOH server (HQ-SW-MOH). Under Multicast Audio Source

Information activate the Enable Multicast Audio Sources on this MOH Server checkbox. Click Save.

Step 67 Verify and set the following parameters under Multicast Audio Source Information: Base Multicast IP Address: 239.1.1.1 Base Multicast Port Number: 16384 Increment Multicast on: IP Address Max Hops: 4

Step 68 Save and reset the MOH server. Step 69 Go to Media Resources > Media Resource Group, click Find, and select the General_mrg

media resource group. Step 70 Activate the Use Multicast for HQ-SW-MOH Audio (If at least one multicast HQ-SW-MOH

resource is available) check-box, click Save and reset the group. Step 71 On your HQ router enable multicast routing (configure the serial sub-interface used in your pod):

ip multicast-routing interface FastEthernet0/0.1 ip pim sparse-dense-mode interface FastEthernet0/0.2 ip pim sparse-dense-mode interface FastEthernet0/0.3 ip pim sparse-dense-mode ! Serial0/0/0.104 connects to BR in pod1: interface serial0/0/0.104 ip pim sparse-dense-mode ! Serial0/0/0.508 connects to BR in pod2: interface serial0/0/0.508 ip pim sparse-dense-mode

Configuration 4-4: On HQ router enable multicast routing

Step 72 On your BR router enable multicast routing (configure the serial sub-interface used in your pod): ip multicast-routing interface FastEthernet0/0.1 ip pim sparse-dense-mode interface FastEthernet0/0.2 ip pim sparse-dense-mode ! Serial0/0/0.401 connects to HQ in pod1: interface serial0/0/0.401 ip pim sparse-dense-mode ! Serial0/0/0.805 connects to HQ in pod2: interface serial0/0/0.805 ip pim sparse-dense-mode

Configuration 4-5: On BR router enable multicast routing

Step 73 Place a call from Phonex-1 to Phonex-2 and at Phonex-1 put the call on hold. Phonex-2 should play MOH. Keep the call open.


Step 74 Check the IP address of Phonex-2 (Settings > Network Configuration). Step 75 On a CIPC-emulated phone (e.g. Phonex-1) open a web browser and connect to the IP address of

Phonex-2. Click the Stream 1 link to see details about the current RTP stream. The Local Address should be 239.1.1.1/16384 which indicates that the phone listens to the multicast MOH stream.

Step 76 Place a call from Phonex-1 to Phonex-3 and at Phonex-1 put the call on hold. Phonex-3 will play tone on hold only. Phonex-3 does not play MOH because the MOH server streams by default G.711 only. Before changing to multicast MOH, Phonex-3 played MOH because of the transcoder media resource, but multicast streams cannot be transcoded.

Note MOH server could be configured to stream the G.729 codec, (by using the Supported MOH Codecs service parameter of the Cisco IP Voice Media Streaming App service). Using G.729 for MOH, however, is not recommended because the G.729 codec audio quality for music is poor; G.729 is designed and optimized for human speech, and does not work well with music.

Enable Multicast MOH to Branch

CUCM has to instruct the phone to listen to a G.711 MOH stream. Consequently, G.711 must be enabled between BR phones and the MOH server in region configuration. At this point the MOH server shares the same region with all other HQ devices. In order to limit calls to these other devices to G.729 but allow G.711 between the MOH server and the BR phones, the MOH server needs to be placed into a separate, dedicated region.

Step 77 In CUCM Administration go to System > Region and click Add New. Step 78 Create a new region named MOH and click Save. Step 79 Using the Modify Relationship to other Regions pane allow the G.711 Audio Codec to be used to

region HQ by highlighting region HQ in the Regions list and selecting G.711 in the Audio Codec drop-down list. Click Save.

Note You have to click Save after each change in the Modify Relationship to other Regions pane. The changes will then appear in the Region Relationships pane.

Step 80 Using the same technique, allow the G.711 Audio Codec for calls between region MOH and region BR and for calls within region MOH. Allow G.729 only for calls between region MOH and Trunks.

Note By limiting the audio codec to G.729 for calls between regions Trunks and MOH you effectively disable MOH for these calls. The MOH server is only streaming G.711 multicast MOH and a multicast stream cannot be transcoded (which would be required towards region Trunks). This is desired as G.729 MOH has only poor quality and G.711 must not be sent over the IP WAN (used by the trunks).

Step 81 Go to System > Device Pool, click Add New and create a new device pool with these parameters:

Device Pool Name: MOH_Only Cisco Unified Communications Manager Group: Default Date/Time Group: CMLocal Region: MOH SRST Reference: Disable


Step 82 Go to Media Resources > Music On Hold Server, click Find, and select HQ-SW-MOH. Step 83 Change the device pool from Default to MOH_Only, click Save, and reset the MOH server.

Verify that Multicast MOH now works to BR Phones

Step 84 Place a call from Phonex-1 to Phonex-3 and at Phonex-1 put the call on hold. Phonex-3 should play MOH.
















CIPT260

Implementing Call Admission Control

1. Objective In this exercise you will first configure locations-based CAC for calls between headquarters, branch, SIP ITSP, and the H.225 trunk. Then you will change the previously implemented locations-based CAC to use RSVP in the IP WAN for calls between the headquarters and branch locations by deploying RSVP agents at each site.

Next you will configure a backup path for calls that are rejected by the previously implemented CAC methods. These calls will be re-routed over the PSTN using Automated Alternate Routing (AAR) and Call Forward No Bandwidth (CFNB).

Finally you will configure a gatekeeper to perform CAC for intercluster calls using an H.225 trunk and verify that the PSTN is used as a backup in case that the gatekeeper rejects the intercluster calls.

































Parameter Value



10.x.5.0/24


10.x.4.0/24




PSTN LAN 10.3.0.0/24





4. Detailed Instructions Follow the steps in the tasks to implement call admission control.

Task 1: Configure Locations In this task you will configure locations-based CAC for calls between headquarters, branch, SIP ITSP, and the H.225 trunk.

Step 1 Connect to your Phonex-1 (x is your pod number) and login in as administrator with password admin. Access CUCM Administration by clicking its desktop shortcut (https://10.x.1.1/ccmadmin). Log in as CCMAdministrator with password appuserpass. Update the MAC addresses of your classroom phones as needed.

Note Alternatively you could also access CUCM Administration from a PC plugged to Phonex-1 in the classroom.

Step 2 In CUCM administration, go to System > Location, click Find and select Hub_None. Make sure that the Hub_None location has unlimited audio bandwidth.

Step 3 Click Add New to create three new locations Branch, ICT and SIP, each with audio bandwidth limited to 24 kpbs. Save them.

Step 4 Go to System > Device Pools, click Find, select Default and set its location to Hub_None. Save and reset the device pool.

Step 5 Use the same procedure to apply the Hub_None location to the device pool MOH_Only and the Branch location to the device pool Branch.

Step 6 Go to Device > Trunk, click Find, select the SIP-Trunk, and set its location to SIP. Save and reset the trunk.

Step 7 Use the same procedure to apply the ICT location to the device pool H225-Trunk_x.

Verification Step 8 Make test calls verifying that calls that exceed the limits are rejected because of not enough

bandwidth:

Note The backup paths going through the PSTN have been deleted to demonstrate AAR.

You should not be able to place more than 1 call to the other cluster using the H.225 trunk (G.729 codec should be in use for this call).


You should not be able to place more than 1 call to the SIP ITSP (e.g. 9-1010-111-555-1234) (G.729 codec should be in use for this call).

Note The CIPC software may crash when attempting an unauthorized call. In that case proceed to the next step.


You should not be able to place more than 1 call to Phonex-3 (G.729 codec should be in use for this call).

Task 2: Configure RSVP-Enabled Locations In this task you will change the previously implemented locations-based CAC to use RSVP in the IP WAN for calls between the Hub_None and Branch locations. This will be done by deploying RSVP agents at the HQ and BR sites.

Step 9 Go to System > Locations, click Find, and select Branch. Step 10 In the Modify Setting(s) to Other Locations pane select the Hub_None location and in the RSVP

Setting menu select Mandatory (Video Desired). Click Save, then Resync Bandwidth button to reset all CAC bandwidth usage, and OK.

Note RSVP is configured per pair of locations. The setting applies to both directions. Therefore the configuration that you apply to one location automatically updates the other location accordingly.

Step 11 On your HQ router, configure MTP resources to act as an RSVP agent: voice-card 0 dspfarm dsp services dspfarm ! sccp local loopback0 sccp ccm 10.x.1.1 identifier 1 version 5.0.1 sccp ! sccp ccm group 1 associate ccm 1 priority 1 associate profile 1 register HQ-RSVP ! dspfarm profile 1 mtp codec pass-through rsvp maximum sessions software 2 associate application SCCP no shutdown ! interface Serial0/0/0 bandwidth 2000 fair-queue interface Serial0/0/0.104 ! link to BR in pod1 ip rsvp bandwidth 24 interface Serial0/0/0.508 ! link to BR in pod2 ip rsvp bandwidth 24

Configuration 4-1: Configure MTP resources to act as an RSVP agent


Step 12 On your BR router, configure MTP resources to act as an RSVP agent:

voice-card 0 dspfarm dsp services dspfarm ! sccp local loopback0 sccp ccm 10.x.1.1 identifier 1 version 5.0.1 sccp ! sccp ccm group 1 associate ccm 1 priority 1 associate profile 1 register BR-RSVP ! dspfarm profile 1 mtp codec pass-through rsvp maximum sessions software 2 associate application SCCP no shutdown ! interface Serial0/0/0 bandwidth 2000 fair-queue interface Serial0/0/0.401 ! link to HQ in pod1 ip rsvp bandwidth 24 interface Serial0/0/0.805 ! link to HQ in pod2 ip rsvp bandwidth 24

Configuration 4-2: Configure MTP resources to act as an RSVP agent

Step 13 In CUCM Administration, go to Media Resources > Media Termination Point and click Add New.

Step 14 Create a new MTP with type ‘Cisco IOS Enhanced Software Media Termination Point’, name HQ-RSVP, description HQ RSVP Agent, and device pool Default. Save it.

Note The name is case sensitive. Location Hub_None and region HQ are applied to the HQ-RSVP media resource through the Device Pool Default.

Step 15 Click Copy and change the name to BR-RSVP, the description to BR RSVP Agent, and the device pool to Branch. Save it.

Note Location Branch and region BR are applied to the HQ-RSVP media resource through the Device Pool Branch.


Verification Step 16 Place a call between an HQ phone and Phonex-3. The call should fail. Step 17 On your HQ router start the debug ip rsvp signalling command to see why the reservation fails.

How much bandwidth do you expect to be reserved? How much is actually tried to be reserved?

Note During the call setup phase the RSVP agents always attempt to reserve additional 16 kbps (for signaling). Therefore, the rsvp bandwidth at the interface must allow 40 kpbs for the call. The extra 16 kbps that are tried to be reserved during call setup are immediately released once the call is set up.

Step 18 Change the rsvp bandwidth at the IP WAN interfaces on the HQ and on the BR routers to 40 kpbs. Now the call should go through. If you want you can retry with 39 kbps to make sure that 40 kbps is the absolute minimum for one G.729 call to be allowed. Make sure that you set it back to 40 kpbs afterwards.

Note The RSVP bandwidth command has to be modified on both sides. Unless both sides permit enough bandwidth for RSVP the call will fail.

Step 19 Establish one call between a HQ phone and Phonex-3 and keep the call open. From the other HQ phone call Phonex-3. The second call should fail.

Step 20 Turn off all debug commands at all routers.

Task 3: Configure AAR and CFNB to Route Calls over the PSTN if they are not admitted by the deployed CAC methods

In this task you will configure a backup path for calls that are rejected by the previously implemented CAC methods. These calls will be re-routed over the PSTN using AAR and CFNB.

Note You will create new route patterns for AAR. One route pattern will be used for the HQ using a new route list that includes its local (HQ) PSTN gateway and one will be used for the BR using a new route list that includes its local (BR) PSTN gateway. The route patterns will be in different partitions. You will create a dedicated CSS for each partition.

Step 21 In CUCM administration, go to System > Service Parameters, choose your CUCM server and select Cisco CallManager service.

Step 22 Locate the Clusterwide Parameters (System - CCM Automated Alternate Routing), set the Automated Alternate Routing Enable parameter to True and save.

Note You will configure two AAR groups: group HQ_AAR should use a dial prefix 9 towards group BR AAR and group BR_AAR should use a dial prefix 9 towards the HQ AAR group.

Step 23 Go to Call Routing > AAR groups and click Add New. Create a new AAR group named HQ_AAR and save it. Do not configure any prefix within HQ_AAR.

Step 24 Click Add New, and create a group named BR_AAR and save it.


Step 25 Configure the following parameters in the ‘Prefix Digits between BR_AAR and other AAR Groups’ pane:

Dial Prefix (From BR_AAR): 91 Dial Prefix (To BR_AAR): 91 Save the AAR group.

Note As you can configure both directions at each AAR group your configuration at AAR group BR_AAR automatically updates the other AAR group accordingly. As the same dial prefix is used in both directions, a single AAR group with prefix 91 within the group would have the same effect.

Step 26 Go to Call Routing > Class of Control > Partition and click Add New. Step 27 Create two partitions with these names and descriptions, and save them:

HQ_AAR, AAR PSTN Access (HQ) BR_AAR, AAR PSTN Access (BR)

Configuration 4-3: Create two partitions

Step 28 Go to Call Routing > Class of Control > Calling Search Space and click Add New. Step 29 Create a CSS named HQ-AAR_css with description ‘AAR Access for HQ’. Add to it the

partition HQ_AAR and save. Step 30 Click Copy and create a CSS named BR-AAR_css with description ‘AAR Access for BR’ that

includes the BR_AAR partition. Step 31 Go to Call Routing > Route/Hunt > Route List and click Add New. Step 32 Create a route list named HQ-AAR_rl with CUCM group Default and save. Step 33 Click Add Route Group and select route group HQ_rg. Click Save and OK.

Note The required digit manipulation is already present at the HQ H.323 gateway.

Step 34 Click Add New and create a route list named BR-AAR_rl with CUCM group Default. Click Save.

Step 35 Click Add Route Group and select route group BR_rg. Enter 51x5553XXX for the Calling Party Transform Mask and select NANP:PreDot from the Discard Digits list. Click Save and then OK.

Step 36 Go to Call Routing > Route/Hunt > Route Pattern and click Add New. Step 37 Create a route pattern with these parameters:

Route Pattern: 9.! Route Partition: HQ_AAR Description: HQ PSTN AAR Gateway/Route List: HQ-AAR_rl Save the route pattern

Note If the called device has CFNB configured the call could be forwarded to any PSTN number. Therefore the route pattern that is used for AAR calls has to include all possible PSTN destinations.


Step 38 Click the Add New to create a route pattern with these parameters, and save: Route Pattern: 9.! Route Partition: BR_AAR Description: BR PSTN AAR Gateway/Route List: BR-AAR_rl

Step 39 Go to Device > Phone and click Find, select Phonex-1 and click Line 1. Step 40 In the AAR Settings pane set the AAR destination mask to 51x5552XXX (where x is your pod

number) and the AAR Group to HQ_AAR. Save it. Step 41 From the Related Links select Configure Device and click Go. Set the AAR CSS to HQ-

AAR_css. Click Save and then OK. Step 42 Use the same procedure to configure Phonex-2 in the same way. Step 43 Use the same procedure to configure Phonex-3 but assign AAR Group BR_AAR and set the

AAR destination mask to 52x5553XXX at the line and assign AAR CSS BR-AAR_css to the phone.

Verification Step 44 Reconfigure the RSVP bandwidth on one of the two routers to a value below 40 kbps. This will

make all calls between BR and HQ fail. Step 45 Try calls between HQ and BR. The calls should be rerouted through the PSTN. The number that

is dialed should be composed of the prefix in the AAR group (91) and the external phone number of the called phone. Verify this by enabling debug isdn q931 at the outgoing PSTN gateway.

Step 46 Change Phonex-1 line settings so that if it cannot be reached due to CAC, the call is forwarded to PSTN number 3335551111 (set AAR destination mask).

Note The prefixes in the AAR mask are also applied when an AAR destination mask is set. Therefore the number has to be entered without the digits that are added by the configured AAR prefix (91).

Step 47 From Phonex-3 call Phonex-1. The call should be sent to the PSTN phone. Step 48 Change the RSVP bandwidth back to 40 on both routers. Step 49 Place one call between HQ and BR and verify that it uses the IP WAN. Keep that call open. Try

calling the BR phone from another phone in the HQ. You should see an incoming call at the BR phone (which is still in a call) coming through the PSTN. You can use debug isdn q931 to verify that calls are using the PSTN path.

Step 50 In the line settings of Phonex-1 return the AAR destination mask to 51x5552XXX (where x is your pod number).


Task 4: Configure Gatekeeper-based CAC for H.225 Trunks In this task you will configure a gatekeeper to perform CAC for intercluster calls using an H.225 trunk. Then you will verify that the PSTN is used as a backup in case that the gatekeeper rejects the intercluster calls.

Step 51 Access the GK router by clicking its icon or connecting via telnet to 192.168.3.1. Step 52 Configure the gatekeeper to allow only one G.729 call to be sent to or from your zone by

entering the following commands (x is your pod number): gatekeeper bandwidth interzone zone podx 16

Configuration 4-4: Configure gatekeeper

Note Configure the gatekeeper bandwidth limitation only for your zone. Do not configure a default limit that would apply to all zones.

Verification Step 53 Place a call from a phone in one pod to a phone in the other pod. This call should work. Place a

second intercluster call. What do you expect to happen? Will the call go through or fail? Why do you get this result? Run the debug ras command at the gatekeeper.

Note The CIPC software may crash when attempting an unauthorized call. In that case proceed to the next step.

Step 54 In order to see the gatekeeper CAC working, change the audio bandwidth of the ICT location to unlimited.

Step 55 Try again to place two intercluster calls. This time you should see an admission request at the gatekeeper also for the second call. However, due to the configured bandwidth limit the gatekeeper will reject the call.

Note For a clear demonstration of CAC, the PSTN backup paths have been deleted.
















CIPT260

Implementing Device Mobility

1. Objective In this exercise you will enable the Cisco Unified Communications Manager device mobility feature to facilitate mobile users that roam away from their home location. First you will configure device mobility in a way that roaming-sensitive settings are applied to phones roaming between different physical locations and between different device mobility groups. This configuration does not modify call routing behavior (i.e. the home gateway is used for PSTN calls); it only adapts to settings such as regions and locations.

Then you will reconfigure device mobility so that roaming-sensitive settings and device mobility-related settings are applied to phones roaming between different physical locations and within the same device mobility group. This configuration modifies call routing behavior (i.e. the gateway located at the physical location is used for PSTN calls) in addition to settings such as regions and locations.




Note To verify device mobility, you need to have classroom phones.





























Parameter Value



10.x.5.0/24


10.x.4.0/24




PSTN LAN 10.3.0.0/24





4. Detailed Instructions Follow the steps in the tasks to implement device mobility.

To verify device mobility, you need to use classroom IP phones, CIPC-emulated phones are not enough.

Task 1: Configure Device Mobility Using Different Device Mobility Groups

In this task you will configure device mobility in a way that roaming-sensitive settings are applied to phones roaming between different physical locations and between different device mobility groups. This configuration does not modify call routing behavior (i.e. the home gateway is used for PSTN calls); it only adapts to settings such as regions and locations.

Step 1 Connect to your Phonex-1 (x is your pod number) and login in as administrator with password admin. Access CUCM Administration by clicking its desktop shortcut (https://10.x.1.1/ccmadmin). Log in as CCMAdministrator with password appuserpass. Update the MAC addresses of your classroom phones.

Note Alternatively you could also access CUCM Administration from a PC plugged to Phonex-1 in the classroom.

Step 2 Go to System > Physical Location and click Add New to create an entry named HQ_pl with description ‘Headquarters’. Click Save.

Step 3 Click Add New to create an entry named BR_pl with description ‘Branch’. Click Save. Step 4 Go to System > Device Mobility > Device Mobility Group and click Add New. Step 5 Create two new entries named DMG-A and DMG-B and save them. Step 6 Go to System > Device Pool, click Find and select device pool Default and configure these

parameters: Physical Location: HQ_pl Device Mobility Group: DMG-A Device Mobility Related Information:

— Device Mobility CSS: HQ-Phones_css — AAR Calling Search Space: HQ-AAR_css — AAR Group: HQ_AAR

Step 7 Save and then reset the device pool. Select Back To Find/List and click Go. Step 8 Select device pool Branch and configure these parameters:

Physical Location: BR_pl Device Mobility Group: DMG-B Device Mobility Related Information:

— Device Mobility CSS: BR-Phones_css — AAR Calling Search Space: BR-AAR_css — AAR Group: BR_AAR

Step 9 Click Save and reset the device pool. Step 10 Go to System > Device Mobility > Device Mobility Info and click Add New.


Step 11 Create a new entry named HQ_dmi for subnet 10.x.2.0/24. Set the device pool to Default and click Save.

Step 12 Click Add New to create a new entry named BR_dmi for subnet 10.x.5.0/24. Set the device pool to Branch and click Save.

Step 13 Navigate to System > Service Parameters, select the Cisco CallManager service, locate the Clusterwide Parameters (Device - Phone) set the Device Mobility Mode to On.

Step 14 Access CUCM Serviceability, go to Tools > Control Center - Feature Services, select the Cisco CallManager service and click Restart.

Verification Step 15 Check that phones can register at different physical locations:

Disconnect Phonex-2 and Phonex-3 from the switch port. Reconnect Phonex-2 to the port of Phonex-3. Reconnect Phonex-3 to the port of Phonex-2. Phonex-2 should register in the branch with its directory number 2002 and Phonex-3 should

register in the headquarters with its directory number 3001. Step 16 Place calls between any of the three phones. Step 17 Check that roaming phones got the roaming sensitive settings updated based on the configuration

in the roaming device pool: Go to Device > Phone, click Find and select Phonex-2. Click the View Current Device Mobility Settings link next to the Device Mobility Mode

parameter. Verify that Phonex-2 adapted to its new physical location (branch) by changing the roaming sensitive settings, such as the location, region, and SRST reference. Note that Phonex-2 did not update its device mobility-related settings, such as AAR group, AAR CSS, and device CSS.

In order to see that these updated settings are active place a call from Phonex-2 to Phonex-1. Press the ? button twice at both phones. The codec used for the call should be G.729.

Note When being in the home location, Phonex-2 uses G.711 for calls to Phonex-1.


Step 18 Test that roaming phones did not get the device mobility-related settings updated: On your HQ and BR routers enable isdn debugging. From Phonex-2 place a call to the PSTN (e.g. 9-333-4444). The call uses the HQ gateway.

This indicates that the device CSS of the phone was not updated by the one configured in the roaming device pool.

Place a call from Phonex-2 to Phonex-3 and keep the call open. At Phonex-2 put the call on hold and try to place a call to Phonex-1. This call should initiate AAR because there is no bandwidth left for calls between the branch and the headquarters. However, as Phonex-2 still uses its home gateway for PSTN calls, AAR fails as Phonex-2 cannot establish a call to the HQ gateway through the overloaded IP WAN.

Note As Phonex-2 was roaming between different device mobility groups, the device CSS and the AAR CSS have not been updated. Codec settings were updated and G.729 is used towards Phonex-1. Also the location is updated; otherwise there would be no problem in placing multiple calls to Phonex-1 (as they were considered being in the same location). As all CSS’s remain unchanged, Phonex-2 is still using its home gateway for PSTN and AAR calls. To allow Phonex-2 to also modify its call routing behavior, the same device mobility group must be configured at the home device pool and the roaming device pool.

Task 2: Configure Device Mobility Using a Single Device Mobility Group

In this task you will configure device mobility in a way that roaming-sensitive settings and device mobility-related settings are applied to phones roaming between different physical locations and within the same device mobility group. This configuration modifies call routing behavior (i.e. the gateway located at the physical location is used for PSTN calls) in addition to settings such as regions and locations.

Step 19 Go to System > Device Pool, click Find and select the device pool Branch. Step 20 Change the Device Mobility Group from DMG-B to DMG-A, click Save and reset.


Verification Step 21 Check that phones can register at different physical locations:

After the reset of the device pool, Phonex-2 should register in the branch with its directory number 2002 and Phonex-3 should register in the headquarters with its directory number 3001.

Place calls between any of the three phones. Step 22 Verify that roaming phones got the roaming sensitive settings and device mobility-related

settings updated based on the configuration in the roaming device pool: Go to Device > Phone, click Find and select Phonex-2. Click the View Current Device Mobility Settings link next to the Device Mobility Mode

parameter. Verify that Phonex-2 adapted to its new physical location (branch) by changing the roaming sensitive settings (such as the location, region, and SRST reference) but also by updating the device mobility-related settings (AAR group, AAR CSS, and device CSS).

In order to see that these updated settings are active place a call from Phonex-2 to Phonex-1. Press the ? button twice at both phones. The codec used for the call should be G.729.

Note When being in the home location, Phonex-2 uses G.711 for calls to Phonex-1.

On HQ and BR router make sure that isdn debugging is active. From Phonex-2 place a call to the PSTN (e.g. 9-333-4444). The call should use the BR

gateway. This indicates that the device CSS of the phone was updated by the one configured in the roaming device pool.

Place a call from Phonex-2 to Phonex-3 and keep the call open. At Phonex-2 put the call on hold and try to place a call to Phonex-1. This call initiates AAR because there is no bandwidth left for calls between the branch and the headquarters. Because of the updated AAR CSS, the BR gateway is be used for the outgoing AAR call and the call is received at Phonex-1 as a PSTN call and can be accepted.

Note As Phonex-2 was roaming within the same device mobility group, the device CSS and the AAR CSS have been updated. If Phonex-2 was configured with a line CSS to implement class of service (by adding partitions to the line CSS that refer to blocked PSTN patterns), the calling privileges would stay intact as the line CSS is not modified by the device mobility feature. Therefore the line/device CSS approach is strictly recommended when using device mobility. It allows to keep class of service (based on the line CSS) combined with a location-depending PSTN gateway selection (based on the device CSS).

Step 23 Put the IP phones back to their initial switch ports.


CIPT260

Implementing Extension Mobility

1. Objective In this exercise you will implement Extension Mobility for roaming users. To achieve this goal, you will first activate the Cisco Extension Mobility (EM) feature service and create the IP phone service required by Extension Mobility. You will configure the login behavior by setting the appropriate service parameter for the extension mobility service and create a device profile for the user of a phone.

Next you will add an end user to Cisco Unified Communications Manager and associate this user with the newly created device profile. Finally you will subscribe phones and the device profile to the extension mobility IP phone service and verify the operations of extension mobility.

































Parameter Value



10.x.5.0/24


10.x.4.0/24




PSTN LAN 10.3.0.0/24





4. Detailed Instructions Follow the steps in the tasks to implement extension mobility.

Task 1: Activate the Required Feature Service and Configure the IP Phone Service

In this task, you will activate the Cisco Extension Mobility (EM) feature service and create the IP phone service required by Extension Mobility.

Step 1 Connect to your Phonex-1 (x is your pod number) and login in as administrator with password admin. Access CUCM Serviceability by clicking its desktop shortcut (https://10.x.1.1/ccmservice). Log in as CCMAdministrator with password appuserpass. Update the MAC addresses of your classroom IP phones as needed.


Step 2 In CUCM Serviceability go to Tools > Service Activation and activate the Cisco Extension Mobility service. Click Save.

Step 3 Access CUCM Administration, go to Device > Device Settings > Phone Services and click Add New.

Step 4 Create a new service named Extension Mobility with these settings, and save it: ASCII Service Name: Extension Mobility Service Description: Extension Mobility Service Service URL (x is your pod number]:

http://10.x.1.1:8080/emapp/EMAppServlet?device=#DEVICENAME#

Task 2: Configure Extension Mobility Service Parameters In this task, you will configure the login behavior by setting the appropriate service parameter for the extension mobility service.

Step 5 Go to System > Service Parameters, choose your CUCM server and select Cisco Extension Mobility.

Step 6 Configure the following parameters, and save them: Enforce Maximum Login Time: True Maximum Login Time: 0:03 Multiple Login Behavior: Auto Logout Remember the Last User Logged In: True Clear Call Log: True


Task 3: Create a Device Profile for a User In this task, you will create a device profile for the user of Phonex-1.

Step 7 Go to Device > Device Settings > Device Profile and click Add New. Step 8 Select Cisco 7940 for the phone model and click Next to create a profile with these parameters:

Note If you use CIPC-emulated phones, select CIPC as the phone model.

Device Profile Name: robin_7940_dp Description: Robin’s Device Profile User Hold MOH Audio Source: 1-Sample Audio Source User Locale: English, United States Phone Button Template: Standard 7940 SCCP Softkey Template: Standard User

Step 9 Click Save and then the Line [1] Add a new DN. Step 10 Set the directory number to 2005, leave the partition blank and click Save.

Task 4: Add and Associate an End User with the User Device Profile

In this task, you will add an end user to Cisco Unified Communications Manager and associate this user with the newly created device profile.

Step 11 Go to User Management > End User and click Add New. Create a user with these parameters: User ID: robin Password: password PIN: 12345 Last name: hood In the Extension Mobility pane select profile robin_7940_dp, move it to the Controlled

Profiles list and save.


Task 5: Subscribe IP Phones and Device Profiles to the Extension Mobility IP Phone Service

In this task, you will subscribe phones and the device profile to the extension mobility IP phone service.

Step 12 Go to Device > Device Settings > Device Profile and select the device profile robin_dp. Step 13 At the Related Links choose Subscribe/Unsubscribe Services; then, click Go. Step 14 Select the Extension Mobility service, click Next, then Subscribe, and close the window. Step 15 Go to Device > Phone, click Find and select Phonex-2. Step 16 In the Extension Information pane activate the Enable Extension Mobility check box. Leave the

default setting of Log Out Profile - Use Current Device Settings. Click Save and then OK. Step 17 At the Related Links choose Subscribe/Unsubscribe Services; then, click Go. Step 18 Select the Extension Mobility service, click Next, then Subscribe, Save, and close the window.

Reset Phonex-2. Step 19 Use the same procedure to enable extension mobility on Phonex-3 and subscribe it to the

extension mobility service.

Verification Step 20 You can login and logout at Phonex-2 and Phonex-3 using this procedure:

Press the Services button. Select the Extension Mobility service. Login with username and PIN. The phone will reset and load with your device profile.


Step 21 Place calls to internal destinations.

Note For this exercise, multisite dial plan and privileges have been deleted from the CUCM’s. However, mobility does not modify device level settings such as region and location or device CSS and AAR CSS. These parameters are not configurable at the device profile. The line CSS configured at the line of the device profile is applied. This means, that when using the line/device CSS approach, gateway selection is based on the actual phone where the user logs in (because of the device CSS) and class of service is based on the device profile of the user (because of the device profile’s line CSS).


Step 22 Verify that the configured service parameters are working: Login at Phonex-2 and place a call. Then wait for the maximum login timer to expire (3

minutes). You should be automatically logged out when the timer expires. Login again at Phonex-2. Verify that the call list was cleared after logout. Press the redial

softkey to verify that the phone does not remember the last destination. Login at Phonex-3 before the 3 minutes timer expires. Once you logged in at Phonex-3 you

should be automatically logged out at Phonex-2 because the multiple login behavior has been set to auto-logout.

After logging out or being logged out of a phone the phone reconfigures itself to its standard settings.

Task 6 (Optional): Configure Extension Mobility for Phonex-1 Step 23 Using the above procedure create a device profile for IP Phone 7961 (robin_7961_dp), set its DN

to 2005, associate the user robin with the new device profile, subscribe Phonex-1 and the new device profile to extension mobility and check its operations by logging in on Phonex-1 as robin with PIN 12345.





Step 25 Click on the phone icon in the topology picture. When you are asked if you want to Save, Run or Cancel the program, select Save. Change name from RDPconnection.rdp to a meaningful name for that phone.










CIPT260

Implementing Cisco Unified Mobility

1. Objective In this exercise you will implement Mobile Connect and Mobile Voice Access.

To implement Mobile Connect, you will configure a softkey template to include the Mobility softkey and apply the softkey template to an IP phone. You will update an existing end user account for Cisco Unified Mobility and associate the user with an office phone. Then you will create Remote Destination Profiles and Remote Destinations. The remote destination profile is a virtual phone that shares its line with the office phone of the user. The remote destination profile represents the associated remote destinations (i.e. PSTN numbers such as mobile phone or home phone).

To implement Mobile Voice Access, you will activate the Cisco Unified Mobile Voice Access Service feature service and configure Cisco Unified Mobility service parameters to enable the Cisco Unified Mobile Voice Access feature globally. Then you will allow individual end users to use Cisco Unified Mobile Voice Access, configure the Cisco Unified Mobility media resources used by mobile voice access, and configure the Cisco IOS Gateway with a call application that allows PSTN calls to be placed from the remote phone as if they were originated from the office phone.

































Parameter Value



10.x.5.0/24


10.x.4.0/24




PSTN LAN 10.3.0.0/24





4. Detailed Instructions Follow the steps in the tasks to implement Cisco Unified Mobility.

Task 1: Add the Mobility Softkey to IP Phones In this task, you will configure a softkey template to include the Mobility softkey and apply the softkey template to an IP phone.

Step 1 Connect to your Phonex-1 (x is your pod number) and login in as administrator with password admin. Access CUCM Administration by clicking its desktop shortcut (https://10.x.1.1/ccmadmin). Log in as CCMAdministrator with password appuserpass. Update the MAC addresses of the classroom phones as needed.


Step 2 Go to Device > Device Settings > Softkey Template and click Find. Locate and click the Copy icon to the right of the Standard User.

Step 3 Enter the name Standard User Mobility, description ‘Mobility Softkey Template’ and click Save.

Step 4 At the Related Links select Configure Softkey Layout and click Go. Step 5 In the On Hook state to configure, click Mobility in the left pane and move it to the right using

the arrow link. Save the setting. Step 6 Repeat the same procedure to add the Mobility softkey to the Connected call state. Step 7 Go to Device > Phone, click Find and select Phonex-1. Step 8 Select Standard User Mobility from the Softkey Template list, click Save, then OK and reset

the phone.

Verification Step 9 On Phonex-1, press the Mobility Softkey. The error message “You are not a valid Mobile User”

should be displayed.


Task 2: Associate an End User Account with the IP Phone and Enable the Use of Mobility

In this task, you will configure an existing end user account for Cisco Unified Mobility and associate the user with an office phone.

Step 10 Navigate to the User Management > End User and click Add New. Step 11 Create a user with ID bob, password password, pin 12345, and some last name. Step 12 In the Mobility Information area of the user, check Enable Mobility check box. Change the

Remote Destination Limit to 1 and click Save. Step 13 Click Device Association, then Find, select Phonex-1 and click Save Selected/Changes. Step 14 Go to Device > Phone, click Find, and select Phonex-1. Step 15 In the Device Information pane select bob from the Owner User ID list. Click Save, OK and

reset the phone.

Verification Step 16 On Phonex-1 press the Mobility Softkey. The “No Mobile Remote Destination found” error

message should be displayed.

Task 3: Configure Remote Destination Profiles and Remote Destinations

In this task, you will configure Remote Destination Profiles and Remote Destinations. The remote destination profile is a virtual phone that shares its line with the office phone of the user. The remote destination profile represents the associated remote destinations (i.e. PSTN numbers such as mobile phone or home phone).

Step 17 Navigate to Device > Device Settings > Remote Destination Profile and click Add New to create a profile with these parameters:

Name: bob-rdp User ID: bob Device Pool: Default Calling Search Space: HQ_Phones_css Privacy: On Rerouting Calling Search Space: HQ_Phones_css Ignore Presentation Indicators (internal calls only): yes (checked)

Note The Rerouting CSS is used for ringing the remote destination when a call is received at the office phone. It is also used for handing calls that are active at the office phone over to a remote destination. The Calling Search Space is the device CSS of the virtual phone representing the remote destinations. In other words, this CSS is used when placing outgoing enterprise calls from a remote destination. This feature, Mobile Voice Access, is used in tasks 4 to 6.


Step 18 Click Save and then click Line [1] - Add a new DN. Enter 2001 for the Directory Number. Set the partition to HQ_Phones and click Save.

Step 19 At the Related Links select Configure Device (bob_rdp) and click Go. Step 20 Click Add a New Remote Destination and create a destination named bob_home with the

destination number 916665554444, set the Delay Before Ringing Timer to 0, and activate the Mobile Phone check box.

Note This parameter allows or disallows calls that are active at the office phone to be handed over to the remote destination.

Step 21 Verify that Enable Mobile Connect option is checked, click Save and then OK.

Note This parameter allows or disallows the remote destination to ring when a call is received at the office phone.

Step 22 In the Association Information pane check the option Line [1] - 2001 in HQ_Phones. Click Save and then OK.

Verification Step 23 Verify that both the office phone and the PSTN phone ring when internal calls are made to

Phonex-1: From Phonex-2 make a call to 2001. The call should be presented to Phonex-1 and to the

PSTN phone line 2 (10digits). Answer the call on the PSTN phone.


View the line 1 button on Phonex-1. Note the color is red, indicating that a call is active at a remote destination on the shared line 2001.

Hand the call over to the office phone by ending the call at the PSTN phone and pressing the Resume softkey on Phonex-1.End the call.

Step 24 Verify that calls from the PSTN phone are presented as calls from the office phone when calling internal directory numbers:

At the PSTN phone, press the LD button to place a call with a long distance calling number and dial 151x5552002.

Verify that the call is presented with the internal number of Phonex-1 (2001) at the receiving phone (Phonex-2). While Phonex-2 is ringing look at the line 1 button on Phone1-x. Note that the color is red, indicating that the remote destination has a call. End the call on the PSTN phone.


Step 25 Verify that both the office and the mobile phones ring when external calls are made to Phonex-1: Ask the students in the other pod to place a call from their PSTN phone line 1 to

151x5552001. The call should be presented to Phonex-1 and PSTN-Phonex line 2 (if not, drop and repeat

the call). The calling number shown at Phonex-1 is the number of the actual caller. The calling number shown at the PSTN phone is 5115552001. (Cisco Unified Mobility preserves the incoming calling party number on the outgoing call to the remote destination but the digit manipulation configured at the H.323 HQ gateway replaces the calling party number with the number of the attendant if the calling party number is different than 2XXX). End the call.

Note In many countries you are not allowed to set a calling party number on outgoing PSTN calls which is different from your actual PSTN number. Therefore the preservation of the calling party number for mobile connect calls depends on the legal regulations or policies of the PSTN provider. As mentioned, in our lab we are actively ensuring that the calling party number of outgoing PSTN calls is never set to a different number than assigned to the ISDN interface (in this case 51x5552XXX).

Step 26 Verify that the office phone can hand an answered call over to the PSTN phone: From Phonex-2 make a call to 2001. The call should be presented to Phonex-1 and PSTN

phone line 2. Answer the call at Phonex-1. Use More softkey twice and then Mobility softkey. Press Select softkey to send the call to

the PSTN phone. Answer the call on the PSTN phone. Keep the call active between Phonex-2 and the PSTN phone and make a call from Phonex-1

to 3001, and answer the call on Phonex-3. Phonex-2 which is connected to the PSTN phone and Phonex-3 which is connected to

Phonex-1 should both show a connection with 2002. Terminate all calls.

Task 4: Enable Cisco Unified Mobile Voice Access In this task, you will activate the Cisco Unified Mobile Voice Access Service feature service. You will configure Cisco Unified Mobility service parameters to enable the Cisco Unified Mobile Voice Access feature globally and then you will allow individual end users to use Cisco Unified Mobile Voice Access.

Step 27 Go to Cisco Unified Serviceability, choose Tools > Service Activation and start the Cisco Unified Mobile Voice Access Service.

Step 28 In CUCM Administration, navigate to System > Service Parameters, select server 10.x.1.1 and Cisco CallManager service.

Step 29 Locate the Clusterwide Parameters (System – Mobility) section and set the Enable Enterprise Feature Access and Enable Mobile Voice Access parameters to True. Save the settings.

Note Enterprise Feature Access allows Cisco Unified Communications Manager features such hold, resume, transfer and conference to be controlled from a remote phone by using DTMF tones.

Step 30 Navigate to the User Management > End User, click Find and select user bob. Step 31 In the Mobility Information area, activate the Enable Mobile Voice Access check box and save.


Task 5: Configure Cisco Unified Mobility Media Resources In this task, you will configure the Cisco Unified Mobility media resources used by mobile voice access.

Step 32 Go to Media Resources > Mobile Voice Access, and configure these parameters: Mobile Voice Access Directory Number: 2999 Partition: HQ_Phones Locale: English United States (select in Available Locales and move it to the Selected

Locales using the arrow link) Step 33 Save the configuration.

Task 6: Configure the Cisco IOS Gateway for Cisco Unified Mobility

In this task, you will configure the Cisco IOS Gateway with a call application that allows PSTN calls to be placed from the remote phone as if they were originated from the office phone.

Step 34 On your HQ gateway configure the IVR Application: application service MVA http://10.x.1.1:8080/ccmivr/pages/IVRMainpage.vxml exit

Configuration 4-1: On HQ gateway configure IVR application

Step 35 On HQ router, configure an incoming POTS dial-peer for the Mobile Voice Access number (51x5552999) and associate the IVR call application with it:

dial-peer voice 29990 pots service MVA incoming called-number 2999 direct-inward-dial exit

Configuration 4-2: On HQ router configure incoming POST dial peer

Note The calls are received with a called party number of 10 digits. However, voice translation profiles are already in place (applied to the voice port) which change the called party E.164 PSTN number to a 4 digit directory number. Therefore, the incoming called-number command is configured with a 4 digit number.


Step 36 On HQ router, configure a VoIP dial-peer to enable the call application running in IOS to contact the MVA media resource in CUCM.

dial-peer voice 29991 voip destination-pattern 2999 session target ipv4:10.x.1.1 dtmf-relay h245-alphanumeric codec g711ulaw no vad

Configuration 4-3: On HQ router configure VoIP dial peer

Note The destination pattern has to match the Mobile Voice Access Directory Number configured at the Mobile Voice Access media resource. It does not have to match the last digits of the PSTN number that is used for mobile voice access (51x5552999 in this case).

Verification Step 37 An outgoing PSTN call can be placed from the remote phone appearing to be initiated from the

office phone: From the PSTN phone line 2 dial 151x5552999. Listen to the IVR script prompt. The remote

destination number (PSTN 916665554444) is recognized and only the PIN is requested from the IVR script.

Enter the PIN 12345 followed by # when prompted by the IVR script. Listen to the IVR script prompt.

Select option 1 to initiate a call from the remote phone to a PSTN destination looking like a call being placed from the office phone.

Enter a PSTN directory number as it would be entered from Phonex-1 (e.g. 911) followed by #.

Verify that the incoming call received at the PSTN phone presents the full directory number of Phonex-1. The call should be received at the PSTN phone at line 5, the emergency line.

End the call. Step 38 Make a call from a non-remote destination and verify that the remote destination is unknown:

From the PSTN phone line 1 dial 151x5552999. The IVR script should now prompt for the remote destination. Enter your remote destination

916665554444# and then your PIN followed by # as prompted by the script. Then press 1 and place a call to a PSTN destination (e.g. 911#). The call should reach the

PSTN ‘Emergency’ line with the calling party number 51x5552001 again. End the call.

Step 39 Make a call from a non-remote destination to an internal phone number: From the PSTN phone line 1 dial 151x5552999. Login by entering your remote destination number (916665554444#) and PIN (followed by

#). Press 1 and place a call to an internal phone (2002# or 3001#). Verify that the call is received at the correct phone with the calling number of Phonex-1

internal directory number (2001). End the call.


Step 40 Dial the Mobile Voice Access number again and activate (option 2) and deactivate (option 3) Mobile Connect capabilities via the TUI:

From the PSTN phone line 2 dial 1151x5552999. Login by specifying your PIN (followed by #) and select option 3 to disable mobility for this

remote location. End the call. From Phonex-2 call 2001. Note that the call is not sent to the remote destination. End the

call. From the PSTN phone line 2 dial 1151x5552999. Login by specifying your PIN (followed by #) and select option 2 to re-enable mobility for

this remote location. End the call. From Phonex-2 call 2001. Note that the call is sent to the remote destination. End the call.

Step 41 Dial the Mobile Voice Access number and use Enterprise Feature Access to place a call on hold and to resume the call:

From the PSTN phone line 2, dial 151x5552999. Login by specifying your PIN (followed by #).

Make a call to an internal phone (e.g. Phonex-3). Accept the call at Phonex-3. Dial the Feature Access Code *81 to place the call on hold. Phonex-3 should beep. At

Phonex-1 the display indicates that the call at the remote phone has been put on hold. Use the Feature Access Code *83 to resume the call.
















CIPT260

Implementing Security in Cisco Unified Communications Manager

1. Objective You will start this exercise by exploiting the vulnerabilities of an unsecured IPT environment. You will use the Wireshark sniffer running on a PC attached to a phone to capture a telephone conversation. You will extract the real-time data and play back the conversation.

You will begin securing the communications by activating the Cisco CTL Provider and the Cisco Certificate Authority Proxy Function services on the CUCM. Next you will install the Cisco CTL client application on a PC, and use the Cisco CTL client to set the cluster to mixed security mode, add security tokens, and sign the CTL.

You will generate LSCs using CAPF and install them in the IP phone. Then you will configure the device security mode of the IP phones to support authenticated calls or encrypted calls and verify the secure communications.

































Parameter Value



10.x.5.0/24


10.x.4.0/24




PSTN LAN 10.3.0.0/24





4. Detailed Instructions Follow the steps in the tasks to implement security.

Task 1 (Optional): Record and Play Back an IP Telephony Conversation

You will use the Wireshark sniffer that is installed on your local PC, which is attached to Phonex-1, to capture a telephone conversation. Then you will extract the real-time data and play back the conversation.

Step 1 From the local PC attached to classroom Phonex-1, access the CUCM Administration, accept the security warning and log in as CCMAdministrator with password appuserpass. Update the MAC addresses of your classroom IP phones as needed.

Note You can access CUCM administration either from your local PC attached to Phonex-1 (https://10.x.1.1/ccmadmin) or by entering NIL e-learning page (http://e-learning.nil.si, logging in, clicking your Phonex-1 computer, and using the desktop shortcut). If the e-learning page is inaccessible, check if the DNS settings obtained via DHCP work in your classroom.

Step 2 On your local PC attached to classroom Phonex-1, launch the Wireshark application and click Capture > Interfaces. From the list of interfaces, click Capture at the one that has an IP address of the data network (10.x.3.0).

Step 3 Place a call from Phonex-1 to Phonex-2, have a short two-way conversation, and hang up the call. In Wireshark stop the live capture.

Step 4 Search for RTP packets. If there are none, search for the first UDP packet that has its source and destination IP address in the HQ phone network (10.x.2.0). Right-click it and choose Decode as. Decode the packets as RTP and click OK to display the relevant UDP packets as RTP data.

Step 5 Identify the RTP streams by clicking one of the RTP packets, then click Statistics > RTP > Show All Streams. When the new window opens, choose the first entry and then click Analyze.

Step 6 An RTP stream analysis window opens. Click the Save Payload… button, choose the .au format for the forward channel, and save the audio data as C:\audio.au. Close Wireshark.

Step 7 Double-click the newly created file (C:\audio.au) from Microsoft Windows Explorer. Microsoft Windows Media Player will play the file, and you can listen to the captured conversation.

Task 2: Activate Services Required for Security In this task, you will activate the Cisco CTL Provider and the Cisco Certificate Authority Proxy Function services.

Step 8 Access CUCM Serviceability (https://10.x.1.1/ccmservice or Navigation drop-down list). Choose Tools > Service Activation.

Step 9 Check the Cisco CTL Provider and Cisco Certificate Authority Proxy Function services and click Save. A message indicating that service activation can take some time appears. Confirm the message by clicking OK and wait until you see an updated window with “Status: Update completed.”


Task 3: Install the Cisco CTL Client In this task you will install the Cisco CTL client application on your local PC attached to Phonex-1.

Step 10 On your local PC attached to classroom Phonex-1, choose Start > Control Panel > Administrative Tools and double-click Services.

Step 11 Select Smart Card and set the startup type to Automatic and start the service, if not yet done so.

Note The above procedure might slightly differ depending on the used version of the Microsoft Windows operating system.

Step 12 From your local PC plugged to classroom Phonex-1, access CUCM administration (https://10.x.1.1/ccmadmin). Choose Application > Plugins and click Find.

Step 13 Locate the Cisco CTL Client plug-in and click the Download link located at the left end of the line. Choose to start the installation and install the CTL client using default settings.

Task 4: Enable Security Using the Cisco CTL Client In this task, you will use the Cisco CTL client to set the cluster to mixed security mode, add security tokens, and sign the CTL.

Step 14 On your local PC plugged to classroom Phonex-1, double-click the Cisco CTL Client shortcut. Step 15 In the CUCM Server window, enter the IP address of your CUCM (10.x.1.1). Enter the username

CCMAdministrator and password appuserpass and click Next. Step 16 Click Set Communications Manager Cluster to Secure Mode and then Next. Step 17 You are prompted to insert a security token. Insert the first security token into a USB port on

your PC and click OK. Step 18 In the Security Token Information window, click Add. Step 19 In the CTL Entries window, click Add Tokens. Step 20 When prompted, remove the first security token, then insert the second token into the USB port

and click OK. Step 21 In the Security Token Information window, click Add. Step 22 In the CTL Entries window verify that your CTL entries list contains the CAPF, your CUCM

server and the Cisco TFTP server (in one line), and two security tokens. Click Finish. Step 23 You are prompted for a password that allows accessing the security token. Ask your instructor

what password to use. You might be prompted to change the password. Again, ask your instructor for the old password and also ask for the new password that you should enter. Click OK to allow the Cisco CTL client software to sign the CTL file using the currently inserted security token.

Note If you do not know what password to use, do not try to guess the password. If you enter an invalid password multiple times, the security token is locked. There is no way to recover a locked security token. It has to be replaced by a new one. Take care when changing the password (if requested to do so) to use exactly the password that your instructor told you to use.


Step 24 When the CTL file has been signed, you see a window that displays the server, file location, and status of the CTL file on your server. A note at the bottom of the window tells you to reload your Cisco Unified Communications Manager server. Click Done to close the Cisco CTL client application.

Step 25 Select Cisco Unified OS Administration in the Navigation drop-down list and click Go. Log in as admin with password adminpass.

Step 26 Choose Settings > Version and click Restart.

Verification Step 27 After the server restarts (a few minutes), access CUCM administration and click System >

Enterprise Parameters. Locate the Cluster security mode setting and verify that its value is 1.

Note Use Ctrl + f combination to search for a string.

Step 28 At each IP phone, press the Settings button. Scroll to the Security Configuration menu and press the Select softkey. Scroll to the CTL File item and view it.

Task 5: Generate Locally Significant Certificates with CAPF In this task, you will generate LSCs using CAPF and install them in the IP phone.

Issuing an LSC to Phone1 Using the Existing MIC for LSC Enrollment Authentication

Step 29 In CUCM administration, choose Device > Phone and click Find. Step 30 Select Phonex-1 and locate the Certificate Authority Proxy Function (CAPF) Information area. Step 31 Set the Certificate Operation field to Install/Upgrade and the Authentication Mode to By

Existing Certificate (precedence to MIC).

Note If you use CIPC-emulated phones, use the enrollment procedure of Phone2 or Phone3, as CIPC does not have a MIC.

Step 32 Click Save and reset the IP phone.

Note After the phone reregistered, it will generate RSA keys and then request a certificate from CAPF. This request will be authenticated by its existing MIC. You can watch this process by pressing the Settings button of the phone, scrolling to the Security Configuration option, pressing Select, and then viewing the LSC item. The phone screen displays enrollment status information. When the LSC status is “Installed,” the process has finished. Click Exit twice.


Issuing an LSC to Phone2 Using an Authentication String for LSC Enrollment

Step 33 In Related Links go to Back to Find/List. Select Phonex-2 and locate the Certificate Authority Proxy Function (CAPF) Information area.

Step 34 Set the Certificate Operation field to Install/Upgrade and the Authentication Mode to By Authentication String.

Step 35 Click Generate String and write down the authentication string: ____________________________

Step 36 Click Save and reset the phone. Step 37 On Phonex-2, press Settings, scroll to the Security Configuration option and press the Select

softkey. Step 38 Unlock the IP phone configuration by pressing **#. Step 39 Scroll to LSC and press the Update softkey. Enter the authentication key and press the Submit

softkey.

Note The IP phone will generate RSA keys and then request a certificate from CAPF. During this process, the phone screen will provide status information. When the signed certificate is installed, the process has finished. Click Exit twice.

Step 40 When the phone registers, verify the status of the LSC by pressing Settings, scrolling to the Security Configuration option, pressing the Select softkey, and scrolling to the LSC option. You should see status “Installed”. Press the Exit softkey twice to get back to the normal IP phone display.

Issuing an LSC to Phone3 Using No Authentication for LSC Enrollment

Step 41 In Related Links go to Back to Find/List. Select Phonex-3 and locate the Certificate Authority Proxy Function (CAPF) Information area.

Step 42 Set the Certificate Operation field to Install/Upgrade and the Authentication Mode to By Null String.

Step 43 Set the Key Size (Bits) to 512. Click Save and reset the phone.

Note After the phone reregisters, it will generate RSA keys and then request a certificate from CAPF. This request will be not be authenticated. You can watch this process by pressing the Settings button of the phone, scrolling to the Security Configuration option, pressing the Select softkey to display the Security Configuration menu, and then scrolling to the LSC menu item. The phone screen then displays enrollment status information. When the LSC status is “Installed,” the process has finished. Click the Exit softkey twice.

Verification Step 44 In Related Links go to Back to Find/List. Step 45 Populate the fields of the Find and List Phones window to find IP phones where the ‘LSC

status’ ‘is exactly’ ‘upgrade success’ and click Find. You should see your three phones in the list.


Task 6: Configure Authentication and Encryption In this task, you will configure the device security mode of the IP phones to support authenticated calls or encrypted calls.

Step 46 In CUCM administration, go to System > Security Profile > Phone Security Profile and click Find.

Step 47 Locate the Cisco 7961 – Standard SCCP Non-Secure Profile and click the copy icon at the right end of the row.

Note If you use CIPC-emulated phones, modify appropriate profile. CIPC supports only authentication, no encryption. It does not support TFTP encrypted config.

Step 48 Change the name to Cisco 7961 – SCCP Authenticated Profile. Step 49 Set the Device Security Mode to Authenticated, activate the TFTP Encrypted Config check

box, set the Authentication Mode to By Existing Certificate (precedence to LSC). Click Save.

Step 50 In the Related Links area at the top right corner of the screen, choose Back to Find/List and click Go.

Step 51 Locate the Cisco 7940 – Standard SCCP Non-Secure Profile and click the copy icon. Step 52 Change the name to Cisco 7940 – SCCP Encrypted Profile. Step 53 Set the Device Security Mode to Encrypted, the Authentication Mode to By Existing

Certificate (precedence to LSC) and click Save. Step 54 Choose Devices > Phones. Select Phonex-1, locate the Protocol Specific Information area and

change the Device Security Profile to Cisco 7961 – SCCP Authenticated Profile. Click Save and reset the phone.

Step 55 In the Related Links area at the top right corner of the screen, choose Back to Find/List and click Go.

Step 56 Select Phonex-2, locate the Protocol Specific Information area and change the Device Security Profile to Cisco 7940 – SCCP Encrypted Profile. Save and reset the phone. Do the same on Phonex-3.

Verification Step 57 Place a call between Phonex-1 and Phonex-2. During the call, the phones should show the

Authentication symbol (a shield) on their screens.


Step 58 Place a call between Phonex-2 and Phonex-3. During the call, the phones should show the Encryption symbol (a closed lock) on their screens.

Step 59 (Optional) Plug your local PC to Phonex-2 instead of Phonex-1. Capture an encrypted conversation between Phonex-2 and Phonex-3 and try to listen to it. You will not hear anything.














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