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Page 1: Ccie Voice Advanced Lab Workbook
Page 2: Ccie Voice Advanced Lab Workbook

CCBOOTCAMP’s

CCIE Voice Advanced Lab Workbook Volume 1

for the CCIE Voice Lab Exam version 3.0

For questions about this workbook please visit: www.voiceie.com

CCBOOTCAMP

375 N. Stephanie Street

Building 21, Suite 2111

Henderson, NV 89014

1.877.654.2243 Toll Free

www.ccbootcamp.com

“Cisco,” the “Cisco Logo,” “CCNA,” “CCNP,” “CCDP,” “CCDA,” “CCIE,” “Cisco Certified

Network Associate,” “Cisco Certified Design Professional,” “Cisco Certified Design

Associate,” “and “Cisco Certified Network Professional,” are registered trademarks of

Cisco Systems, Inc. The contents contained wherein, is not associated or endorsed by

Cisco Systems, Inc.

Page 3: Ccie Voice Advanced Lab Workbook

PLEASE READ THIS SUBSCRIPTION LICENSE AGREEMENT CAREFULLY BEFORE USING THIS PRODUCT.

THIS SUBSCRIPTION LICENSE AGREEMENT APPLIES TO CCBOOTCAMP’s CCIE Voice Advanced Lab

Workbook.

BY ORDERING THIS PRODUCT YOU ARE CONSENTING TO BE BOUND BY THIS LICENSING AGREEMENT.

IF YOU DO NOT AGREE TO ALL OF THE TERMS OF THIS LICENSE, THEN DO NOT PURCHASE THIS

PRODUCT.

License Agreement

CCBOOTCAMP’s CCIE Voice Advanced Lab Workbook is copyrighted. In addition, this

product is at all times the property of CCBOOTCAMP, and the customer shall agree to

use this product only for themselves, the licensed user. The license for the specific

customer remains valid from the purchase date until they pass their CCIE Voice lab

exam.

CCBOOTCAMP’s CCIE Voice Advanced Lab Workbook materials are licensed by individual

customer. This material cannot be resold, transferred, traded, sold, or have the price

shared in any way. Each specific individual customer must have a license to use this

product. The customer agrees that this product is always the property of CCBOOTCAMP,

and they are just purchasing a license to use it. A Customer’s license will be revoked

if they violate this licensing agreement in any way.

Copies of this material in any form or fashion are strictly prohibited. If for any

reason a licensed copy of this material is lost or damaged a new copy will be provided

free of charge, except for the cost of printing, shipping and handling.

Individuals or entities that knowingly violate the terms of this licensing agreement

may be subject to punitive damages that CCBOOTCAMP could seek in civil court. Damages

will be limited to a maximum of $500,000.00 per individual and $2,000,000.00 per

entity. In addition, individuals or entities that knowingly violate the terms of this

license agreement may be subject to criminal penalties as are allowed by law.

The venue of any dispute, controversy, litigation or proceeding (formal or informal)

arising out of or pertaining to this licensing agreement or the subject hereof shall

lie exclusively in the County of Clark, State of Nevada. Provided, however, that if

any such dispute, controversy, litigation or proceeding requires or permits

jurisdiction in a federal court or agency of the United States, then venue shall lie

in no federal court or agency other than those located in (or nearest to) the County

of Clark, State of Nevada.

Term and Termination of License Agreement

This License is effective until terminated. Customer may terminate this License at any

time by destroying all copies of written and electronic material of said product.

Customer's rights under this License will terminate immediately without notice from

CCBOOTCAMP, if Customer fails to comply with any provision of this License. Upon

termination, Customer must destroy all copies of material in its possession or

control. The license for the specific user remains valid from the purchase date until

the user passes their lab exam pertaining to the purchased subscription. Once the

customer passes the relevant lab exam the license is terminated and all material

written or electronic in their possession or control must be destroyed or returned to

CCBOOTCAMP.

Warranty

No warranty of any kind is provided with this product. There are no guarantees that

the use of this product will help a customer pass any exams, tests, or certifications,

or enhance their knowledge in any way. The product is provided on an “AS IS” basis.

In no event will CCBOOTCAMP, its suppliers, or licensed resellers be liable for any

incurred costs, lost revenue, lost profit, lost data, or any other damages regardless

of the theory of liability arising out of use or inability to use this product.

Page 4: Ccie Voice Advanced Lab Workbook

CCBOOTCAMP’s CCIE Voice

Full Lab Workbook Volume 1

Version 3.0

CCIE Voice 3.0 Blueprint

CCBOOTCAMP, Inc.

375 N Stephanie St.

Building 21, Suite 2001

Henderson, NV 89014

702-968-5100

www.ccbootcamp.com

The CCBOOTCAMP CCIE Full Lab Workbook Volume 1 is licensed by

individual user. They can not be resold or have the price shared in any

way. We have invested a significant amount of time and money into

developing these labs.

“Cisco” the “Cisco Logo,” “CCNA,” “CCNP,” “CCDP,” “CCDA,” “CCVP,”

“CCIE,” “Cisco Certified Network Associate,” “Cisco Certified Network

Professional,” “Cisco Certified Design Professional,” “Cisco Certified

Design Associate,” “Cisco Certified Voice Professional,” and “Cisco

Certified Internetwork Expert,” are registered trademarks of Cisco

Systems, Inc. The contents contained wherein, are not associated with

or endorsed by Cisco Systems, Inc.

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Copyright ©2009, Network Learning, Incorporated

Page 5: Ccie Voice Advanced Lab Workbook

Table of Contents:

...............................................A NOTE FROM THE AUTHOR:! 3

..................................................GENERAL INFORMATION:! 4

CCIE VOICE BLUEPRINT ..............................................3.0:! 4

PRE CONFIGURATION .....................................................:! 5

VPN ACCESS INFORMATION .................................................:! 6

TIPS FOR THE INITIAL SETUP ..............................................:! 6

CISCO'S NON-DISCLOSURE AGREEMENT ........................................:! 7

READ BEFORE START .....................................................:! 7

LAB TOPOLOGY ........................................................:! 8

LAB DIAL PLAN AND ADDRESSING ............................................:! 9

....................................................CHAPTER 1 - LAB 1! 10

BRIEF LAB OVERVIEW ...................................................:! 10

BASIC CAMPUS DESIGN ..................................................:! 10

CALLMANAGER AND CALLMANAGER EXPRESS .....................................:! 11

VOICE GATEWAYS AND SIGNALING ...........................................:! 12

CALL ROUTING ........................................................:! 13

CODEC AND CALL ADMISSION CONTROL ........................................:! 16

HIGH AVAILABILITY FEATURES .............................................:! 16

MEDIA RESOURCES MANAGEMENT .............................................:! 17

QOS FEATURES ........................................................:! 18

VOICEMAIL INTEGRATIONS ................................................:! 19

IPCC EXPRESS VOICE APPLICATION .........................................:! 20

CALLMANAGER VOICE APPLICATION ..........................................:! 21

PRESENCE ............................................................:! 22

SOLUTIONS ...........................................................:! 23

...................................................CHAPTER 2 - LAB 2! 192

BRIEF LAB OVERVIEW ..................................................:! 192

BASIC CAMPUS DESIGN .................................................:! 192

CALLMANAGER AND CALLMANAGER EXPRESS ....................................:! 193

VOICE GATEWAYS AND SIGNALING ..........................................:! 195

CALL ROUTING .......................................................:! 195

CODEC AND CALL ADMISSION CONTROL .......................................:! 198

HIGH AVAILABILITY FEATURES ............................................:! 198

MEDIA RESOURCES MANAGEMENT ............................................:! 199

QOS FEATURES .......................................................:! 200

VOICEMAIL INTEGRATIONS ...............................................:! 201

IPCC EXPRESS VOICE APPLICATION ........................................:! 202

CALLMANAGER VOICE APPLICATION .........................................:! 203

PRESENCE ...........................................................:! 203

SOLUTIONS ..........................................................:! 205

...................................................CHAPTER 3 - LAB 3! 376

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Page 6: Ccie Voice Advanced Lab Workbook

BRIEF LAB OVERVIEW ..................................................:! 376

BASIC CAMPUS DESIGN .................................................:! 376

CALLMANAGER AND CALLMANAGER EXPRESS ....................................:! 377

VOICE GATEWAYS AND SIGNALING ..........................................:! 378

CALL ROUTING .......................................................:! 379

CODEC AND CALL ADMISSION CONTROL .......................................:! 382

HIGH AVAILABILITY FEATURES ............................................:! 382

MEDIA RESOURCES MANAGEMENT ............................................:! 383

QOS FEATURES .......................................................:! 384

VOICEMAIL INTEGRATIONS ...............................................:! 384

IPCC EXPRESS VOICE APPLICATION ........................................:! 386

CALLMANAGER VOICE APPLICATION .........................................:! 387

PRESENCE ...........................................................:! 387

SOLUTIONS ..........................................................:! 388

...................................................CHAPTER 4 - LAB 4! 557

BRIEF LAB OVERVIEW ..................................................:! 557

BASIC CAMPUS DESIGN .................................................:! 557

CALLMANAGER AND CALLMANAGER EXPRESS ....................................:! 558

VOICE GATEWAYS AND SIGNALING ..........................................:! 559

CALL ROUTING .......................................................:! 560

CODEC AND CALL ADMISSION CONTROL .......................................:! 564

HIGH AVAILABILITY FEATURES ............................................:! 564

MEDIA RESOURCES MANAGEMENT ............................................:! 564

QOS FEATURES .......................................................:! 565

VOICEMAIL INTEGRATIONS ...............................................:! 566

IPCC EXPRESS VOICE APPLICATION ........................................:! 567

CALLMANAGER VOICE APPLICATION .........................................:! 568

PRESENCE ...........................................................:! 569

SOLUTIONS ..........................................................:! 570

...................................................CHAPTER 5 - LAB 5! 743

BRIEF LAB OVERVIEW ..................................................:! 743

BASIC CAMPUS DESIGN .................................................:! 743

CALLMANAGER AND CALLMANAGER EXPRESS ....................................:! 744

VOICE GATEWAYS AND SIGNALING ..........................................:! 745

CALL ROUTING .......................................................:! 746

CODEC AND CALL ADMISSION CONTROL .......................................:! 749

HIGH AVAILABILITY FEATURES ............................................:! 749

MEDIA RESOURCES MANAGEMENT ............................................:! 750

QOS FEATURES .......................................................:! 751

VOICEMAIL INTEGRATIONS ...............................................:! 751

IPCC EXPRESS VOICE APPLICATION ........................................:! 753

CALLMANAGER VOICE APPLICATION .........................................:! 754

PRESENCE ...........................................................:! 754

SOLUTIONS ..........................................................:! 756

For questions: www.voiceie.comv.a.l.03.09.05.cf.r05.24.06.pages

2

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Copyright ©2009, Network Learning, Incorporated

Page 7: Ccie Voice Advanced Lab Workbook

A note from the Author:

Thank you for choosing CCBOOTCAMP as your partner in your

journey to attain your CCIE Voice certification. Before

beginning this advanced lab workbook, you should have

already completed, or be at the level of, CCBOOTCAMP’s CCIE

Voice Technology workbook.

The advanced practice labs contained in this workbook are

designed to test your speed and knowledge in voice

technology and help you identify the areas where self study

is needed most. This approach will help you attain the

knowledge needed to not only pass the CCIE Voice lab exam,

but also to become an expert in applying voice technology.

In this first advanced lab workbook volume you will be

tested in areas of dial plan, digit manipulation, SIP, IP-

to-IP gateway, and more. Also included, is a lab designed

specifically to test speed and help you gauge if you’re

ready for the actual lab.

You will need to have a complete and in-depth understanding

of the topics contained in this workbook in order to pass

the lab exam.

I wish you the best!

Chris Fortner - CCIE #18065CCIE Voice Program Manager

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Page 8: Ccie Voice Advanced Lab Workbook

General Information:

You may notice that some of the sections within the lab

repeat. This is done on purpose as it is meant to make you

practice the basics over and over again. This will help

your speed and accuracy when taking the lab.

CCIE Voice Blueprint 3.0:

Listed below is the current published 3.0 hardware and

software blueprint from Cisco Systems. The CCBOOTCAMP

remote racks features a hardware, where relevant, and

software match based on the published blueprint.

Lab Equipment:

! Cisco MCS-7845 Media Convergence Servers

! Cisco 3825 Series Integrated Services Routers (ISR)

! Cisco 2821 Series Integrated Services Routers (ISR)

! ISR Modules and Interface Cards

o VWIC2-1MFT-T1/E1

o PVDM2

o HWIC-4ESW-POE

o NME-CUE

! Cisco Catalyst 3750 Series Switches

! IP Phones and Soft Clients

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Page 9: Ccie Voice Advanced Lab Workbook

Software Versions:

Any major software release which has been generally

available for six months is eligible for testing in the

CCIE Voice Lab Exam.

! Cisco Unified Communications Manager 7.0

! Cisco Unified Communications Manager Express 7.0

! Cisco Unified Contact Center Express 7.0

! Cisco Unified Presence 7.0

! Cisco Unity Connection 7.0

! All routers use IOS version 12.4T Train.

! Cisco Catalyst 3750 Series Switches uses 12.2 Main

Train

! Network Interfaces

o Fast Ethernet

o Frame Relay

! Telephony Interfaces

o T1

o E1

Pre Configuration:

The pre configuration files for the voice racks at

CCBOOTCAMP can be downloaded from the link provided below

and will contain the base information for the start of all

labs.

http://www.ccbootcamp.com/download/!Voice/Files-For-NLIs-CCIE-Voice-Tech-

Workbook/V3/configs.zip

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Page 10: Ccie Voice Advanced Lab Workbook

VPN Access Information:

In order to access the voice racks at CCBOOTCAMP you will

first need to download the Cisco VPN client from

www.cisco.com. Once installed you can then download the

appropriate VPN profile from the link provided below for

the rack you have scheduled.

http://www.ccbootcamp.com/download/!Voice/voice-rack-vpn-profiles/

Tips for the initial setup:

! Define under the console (line con 0) the command “no

exec-timeout” in order to avoid inactivity logout.

! Define some useful shortcuts that may save precious

time with aliases. Some common commands: “show call

active voice brief”, “show gatekeeper call status” and

any command you feel might be used heavily throughout

your lab.

! Think ahead and try not to touch any page more than

once or twice.

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Page 11: Ccie Voice Advanced Lab Workbook

Cisco's Non-Disclosure Agreement:

This book was written in high respect for the CCIE

certification and is not intended to violate Cisco's Non-

Disclosure Agreement. We will not answer questions

regarding things that relate to the actual LAB exam and

will not take part in any activities that stand against the

CCIE NDA in any regards. Please visit the link below for

the complete NDA specification.

http://www.cisco.com/web/learning/le3/ccie/exam/violation_rules.html

Read Before Start:

Here are some simple rules to follow during your lab which

should be taken into consideration before you configure any

devices.

1. All devices should register to the subscriber and then

the publisher.

2. All usernames and passwords for devices are “admin” and

“cisco5796” unless specified in your lab.

3. Where you see “YY” that is your rack number.

4. All devices should pull NTP from the HQ1 router and

should be set to the appropriate time zone.

5. Credit in the real lab will only be given for working

solutions. This should be taken into consideration

throughout your lab.

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Page 12: Ccie Voice Advanced Lab Workbook

Lab Topology :

Note: Please note that ports may vary on different racks so

please check your connections with “show cdp neighbors”.

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Page 13: Ccie Voice Advanced Lab Workbook

Lab Dial Plan and Addressing:

Phone Number Protocol Class of Service

HQ Phone 1 1001 SCCP/SIP International

HQ Phone 2 1002 SCCP/SIP Long Distance

BR1 Phone 1 2001 SCCP/SIP International

BR1 Phone 2 2002 SCCP/SIP Internal

BR2 Phone 1 3001 SCCP/SIP International

BR2 Phone 2 3002 SCCP/SIP Local

Site PSTN E.164 Address PSTN Number

HQ 7029461... 7029465000

BR1 7023332... 7023335000

BR2 442076303... 442076305000

ALL 911 911

Server IP Address Login

Publisher 10.1.200.21 admin:cisco5796

Subscriber 10.1.200.25 admin:cisco5796

Unity Connections 10.1.200.22 admin:cisco5796

Presence 10.1.200.23 admin:cisco5796

Contact Center 10.1.200.24 administrator:enableme

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Page 14: Ccie Voice Advanced Lab Workbook

Chapter 1 - Lab 1

Brief Lab Overview:

This lab is designed to test your dial plan knowledge. The

focus is dial plan redundancy and digit manipulation.

Basic Campus Design:

1. Configure VLANS and IP networks in all three sites

according to the following two tables:

!

VLAN HQ BR1 BR2

Servers 1 N/A N/A

Data 20 120 220

Voice 30 130 230

!

Network HQ BR1 BR2

Servers 10.1.200.0/24 N/A N/A

Data 10.YY.20.0/24 10.YY.120.0/24 10.YY.220.0/24

Voice 10.YY.30.0/24 10.YY.130.0/24 10.YY.230.0/24

2. Make sure you set all voice traffic to use the voice

VLAN for all sites.

3. Configure phones on HQ site to have IP addresses

120-130. Use IOS DHCP on the HQ router to accomplish

this task.

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Page 15: Ccie Voice Advanced Lab Workbook

4. Configure BR1 phones to have IP addresses of 130–140

using the IOS DHCP on the BR1 Router.

5. Configure BR2 phones to have IP addresses 210–220

using the IOS DHCP on the BR2 router.

6. Configure HQ router as an NTP master clock with

stratum 2 for local time zone in Los Angeles and

configure BR1 and BR2 to synchronize their clocks to

HQ. BR1 is in New York time zone and BR2 is in India.

CallManager and CallManager Express:

1. Register all HQ and BR1 phones according to the dial

plan table using SCCP loads. BR2 phones should use SIP

loads and register accordingly. Make sure the phones

will display calling names in every scenario presented

(internal and PSTN):

2. Configure directory number 3003 as a second line on

both phones in BR2. When a call comes into 3003 it

should ring on both phones and be answered by the

first user to respond.

3. Make sure you that when the initiator of a conference

call within the BR2 site drops that the conference

call drops.

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Page 16: Ccie Voice Advanced Lab Workbook

4. Make sure all phones display the same general

information aside from the line specific

configuration; this refers to HQ and BR1.

5. Allow BR2 phone 1 to pickup calls, ringing and on

hold, on BR2 phone 2 and vice versa. This should work

for internal and external calls.

6. Both BR2 phone 1 and BR2 phone 2 are members of VIP

sales group and are required to take calls dialed to

442076303111. The calls are expected to ring the

phones in sequential order. Both phones should ring

for 5 seconds and if the call is not answered it

should be forwarded to DN 3000.

Voice Gateways and Signaling:

1. Configure the HQ router as a IOS MGCP gateway using T1

PRI and NI as the line parameters.

2. Configure BR1 router as a IOS H.323 gateway using T1

PRI and NI as the line parameters.

3. Configure BR2 as E1 PRI and make sure all BR2 numbers

are reachable.

4. Configure HQ router as a gatekeeper with the following

details:

Local zone name: voiceie

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Page 17: Ccie Voice Advanced Lab Workbook

Domain name: ccbootcamp.com

Register CallManager and CME to the gatekeeper.

Register the CallManager with a tech-prefix of “1#”

and a zone called “CM”. Register CME with a tech-

prefix of “1#” and to a zone called “CME”.

Call Routing:

1. In every site, phone 1 should be allowed to dial

international and below numbers, phone 2 should only

be allowed to call long distance and below.

2. Configure the following dialing options for users in

HQ:

7+4 digits Calls to BR2

911 and 9911 Emergency

9+7 digits

The first is in the range of

2 to 9

Local

9+1+10 digits

The first and the forth are

in the range of 2 to 9.

Long Distance

9+011+any number of digits

Please allow users to dial

the hash sign at the end of

the number in order to have

faster response

International

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Page 18: Ccie Voice Advanced Lab Workbook

• Local calls from HQ should use local HQ gateway

and BR1 gateway as backup.

• Long distance calls from HQ should use local HQ

gateway as first choice and then BR1 gateway.

• Calls to BR1 PSTN numbers should be routed out

BR1 gateway (Toll Bypass), with fallback to local

HQ gateway.

• International calls to BR2 numbers, using the 7+4

digits pattern, should be routed via the

gatekeeper as VoIP and use the local HQ gateway

as backup.

• International calls to BR2 local numbers should

be routed via the gatekeeper and then via the

local HQ gateway.

• All other International calls should route out

the HQ gateway.

3. Configure the following dialing options for users in

BR1:

7+4 digits Calls to BR2

911 and 9911 Emergency

9+7 digits

The first is in the range of 2

to 9

Local

9+1+10 digits

The first and the forth are in

the range of 2 to 9.

Long Distance

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Page 19: Ccie Voice Advanced Lab Workbook

9+011+any number of digits

Please allow users to dial the

hash sign at the end of the

number in order to have faster

response

International

• Local calls from BR1 should use the local BR1 IOS

gateway and then the HQ gateway as a backup.

• Long distance calls from BR1 should use the BR1

gateway as a first choice and then the HQ gateway

as a backup.

• Calls to HQ PSTN numbers (Area Code+Prefix)

should be routed out the HQ gateway (Toll Bypass)

with fallback to the local BR1 gateway.

• International calls to BR2 numbers, using the 7+4

digits pattern, should be routed via the

gatekeeper as VoIP and use the BR1 gateway as

backup.

• International calls to BR2 local numbers should

be routed via the gatekeeper and then via the

local HQ gateway.

• All other International calls should route out

the BR1 gateway.

4. Configure the following dialing options for users in

BR2:

7+4 digits Calls to HQ and BR1

999 Emergency

9+7 digits Local

9+0+8 digits Long Distance

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Page 20: Ccie Voice Advanced Lab Workbook

9+00+any number of digits

Please allow users to dial

the hash sign at the end of

the number in order to have

faster response

International

• International calls to HQ and BR1 should use the

gatekeeper first and the local PRI as a backup.

Codec and Call Admission Control:

1. All calls within a site should use the G.711 codec.

2. All calls between sites should use the G.729 codec.

3. Allow two concurrent calls to and from BR1.

4. Allow three concurrent calls from HQ and BR1 to BR2.

High Availability Features:

1. Configure BR1 as an SRST router.

2. Make sure you only allow BR1 phone 2 to register via

SRST during a failure.

3. Preserve all PSTN dialing with using 9 as the first

digit in the string.

4. When in SRST mode, the BR1 phone 2 is not allowed to

dial international calls.

5. Allow HQ and BR1 phones to call each other in times

when more then 2 concurrent calls are in process

between the sites.

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Page 21: Ccie Voice Advanced Lab Workbook

Media Resources Management:

1. Configure conferencing for the HQ site to use hardware

based resources in the HQ gateway first then hardware

resources in the BR1 gateway.

2. Configure conferencing for the BR1 site to use

hardware based resources in the BR1 gateway first then

hardware resources in the HQ gateway.

3. Configure transcoding for the HQ site to use hardware

based resources in the HQ gateway first then hardware

resources in the BR1 gateway.

4. Configure transcoding for the BR1 site to use hardware

based resources in the BR1 gateway first then hardware

resources in the HQ gateway.

5. Configure MOH for the HQ site based on unicast and

streamed from the publisher CallManager.

6. Configure multicast MOH for BR1, based on G729 codec

from the subscriber, and streamed from the BR1

gateway.

7. Allow 5 simultaneous meet me conferences for the HQ

users, in the number range 1201-1205. Every meet me

conference should allow a maximum of ten participants.

8. Make sure that when the initiator of an Ad-Hoc

conference hangs up, the conference is allowed to

continue as long as an internal user is still

connected.

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Page 22: Ccie Voice Advanced Lab Workbook

QoS Features:

1. Configure the frame relay link between HQ and BR1 as

if it is 384k in bandwidth, make sure you enable MLP

LFI on it.

• VoIP signaling should have 5% of the line as

guarantied bandwidth.

• VoIP barrier should have 33% of the line as priority

bandwidth.

• All other traffic should be weighted fair queued.

2. Configure the frame relay link between HQ and BR2 as

if it is 1200k in bandwidth.

• VoIP signaling should have 60k of the line reserved

bandwidth.

• VoIP barrier should have 360k of the line priority

bandwidth.

• All other traffic should be weighted fair queued.

3. The BR1 gateway should send the signaling and RTP

traffic marked as CS3 and EF.

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Voicemail Integrations:

1. Configure voicemail for HQ and BR1 based on the Unity

Connections server using the following parameters:

• Pilot 1170, port 1171-1174, MWI on 1180, MWI off

1181.

2. Make sure you prevent Unity Connections from calling

to itself and creating call loops which consume the

available ports.

3. Configure a voicemail box for phones 1 and 2 in HQ and

BR1 with a phone password of 147369 for all users.

4. Configure an auto attendant for HQ with DN 1000. It

should have a customized greeting that allows dialing

an extension and walking through the users directory

by pressing 1. When users press 0 they should be

forwarded to extension 1001. During closed times,

assume normal weekday schedule, it should take a

message and store it in the HQ phone 1 mailbox.

5. Create a second line on BR1 phone 2 with extension

*2002. A call coming to the primary line should roll

to this line if the primary is busy. The user should

be able to use this line for voicemail access. Do not

use any alternate extensions.

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6. Configure BR2 CME to use the local Unity Express as a

voicemail, using the following parameters:

• Pilot 3180.

7. Create voicemail for BR2 phone 1 and 2. Use a blank

phone password for all users.

8. Create an auto attendant for BR2 that will work when

DN 3000 is dialed.

9. Make sure that calls from HQ can route via gatekeeper

and hit the auto attendant in BR2.

IPCC Express Voice Application:

1. Configure the IPCC on the Contact Center server to

work with the CallManager using the following

parameters (all passwords are “cisco” and all pin’s

are “12345”):

Jtapi username: jtapi

Resource manager user: rmjtapi

CTI ports: 1015 – 1019.

2. Configure the auto attendant script to answer when a

call arrives at 1020.

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3. Configure an ICD queue that will be triggered when

dialing extension 1021. HQ phone 1 should be used as

an IP phone agent for user “HQ1” with the password

“cisco”. BR1 phone 1 should be used as a IP phone

agent with user “BR1” with a password “cisco”.

4. Callers should hear music on hold while delayed in

queue.

5. Calls should be routed via the queue based on a skill

named “Sales”. HQ phone 1 should be skilled higher

than BR1 phone 1.

6. Before a call is routed to the queue a check should be

performed to see if any agents are logged into the

queue. If no agents are logged in then the call should

be routed to HQ phone 2.

CallManager Voice Application:

1. Create extension mobility for user “john” with a

password of “cisco”. Which will allow him to use

extension 1007 on HQ and BR1 phone 2 in each site.

Make sure that the extension mobility service is only

available on phone 2 in HQ and BR1.

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Presence:

1. Integrate the Presence server with CallManager. Make

sure the HQ Phone 1 user can login via CUPC and

control the desktop phone. HQ Phone 1 user should also

be allowed to switch to softphone mode.

2. Make sure HQ Phone 1 can see the voicemail in their

voicemail box through the CUPC client.

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Solutions:

Basic Campus Design

1. Start with “show vlan” and make sure you have all

VLANS defined in all devices. If not create them and

then add the IP interfaces to each device with the

correct IP subnet from the table provided:

Time Saver - In order to save time you should go ahead and configure the

layer 2 QoS in the HQ switch.

HQ:

HQ-3750#

HQ-3750#config t

Enter configuration commands, one per line. End with CNTL/Z.

HQ-3750(config)#vlan 20!! !

HQ-3750(config-vlan)#name Data

HQ-3750(config-vlan)#exit

HQ-3750(config)#vlan 30!! !

HQ-3750(config-vlan)#name Voice

HQ-3750(config-vlan)#exit

HQ-3750(config)#

HQ-3750(config)#interface range FastEthernet 1/0/6 - 7

HQ-3750(config-if-range)#switchport voice vlan 30

HQ-3750(config-if-range)#switchport access vlan 20

HQ-3750(config-if-range)#switchport trunk encapsulation dot1q

HQ-3750(config-if-range)#switchport mode trunk

HQ-3750(config-if-range)#auto qos voip cisco-phone

HQ-3750(config)#interface range FastEthernet 1/0/1 - 5

HQ-3750(config-if-range)#switchport host

HQ-3750(config-if-range)#auto qos voip trust

HQ-3750(config)#interface FastEthernet 1/0/24

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HQ-3750(config-if)#auto qos voip trust

HQ-3750(config)#interface FastEthernet0/0.20

HQ-3750(config-if)#encapsulation dot1Q 20

HQ-3750(config-if)#ip address 10.1.20.1 255.255.255.0

HQ-3750(config)#exit

HQ-3750(config)#interface FastEthernet0/0.30

HQ-3750(config-if)#encapsulation dot1Q 30

HQ-3750(config-if)#ip address 10.1.30.1 255.255.255.0

! Tip - Don’t forget to add the new networks into the routing protocol.

HQ-3750(config)#router ospf 1

HQ-3750(config-router)#network 10.1.20.0 0.0.0.255 area 0

HQ-3750(config-router)#network 10.1.30.0 0.0.0.255 area 0

BR1:

BR1#vlan database

BR1(vlan)#vlan 120 name Data state active

VLAN 120 added:

Name: Data

State ACTIVE

BR1(vlan)#vlan 130 name Voice state active

VLAN 130 added:

Name: Voice

State ACTIVE

BR1(vlan)#exit

APPLY completed.

Exiting....

! Tip - The “vlan database” command has been deprecated so you can now use

the “vlan” command under global configuration mode as well.

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BR1#config t

Enter configuration commands, one per line. End with CNTL/Z.

BR1(config)#int vlan 120

BR1(config-if)#ip address 10.1.120.1 255.255.255.0

BR1(config-if)#description Data

BR1(config-if)#int vlan 130

BR1(config-if)#ip address 10.1.130.1 255.255.255.0

BR1(config-if)#description Voice

BR1(config-if)#exit

BR1(config)#interface range FastEthernet 0/3/0 - 1

BR1(config-if-range)#switchport voice vlan 130

BR1(config-if-range)#switchport access vlan 120

BR1(config-if-range)#no shut

! Tip - Don’t forget to add the new networks into the routing protocol.

BR1(config)#router ospf 1

BR1(config-router)#network 10.1.120.0 0.0.0.255 area 0

BR1(config-router)#network 10.1.130.0 0.0.0.255 area 0

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BR2:

BR2#vlan database

BR2(vlan)#vlan 220 name Data state active

VLAN 220 added:

Name: Data

State ACTIVE

BR2(vlan)#vlan 230 name Voice state active

VLAN 230 added:

Name: Voice

State ACTIVE

BR2(vlan)#exit

APPLY completed.

Exiting....

BR2#config t

Enter configuration commands, one per line. End with CNTL/Z.

BR2(config)#int vlan 220

BR2(config-if)#ip address 10.1.220.1 255.255.255.0

BR2(config-if)#description Data

BR2(config-if)#int vlan 230

BR2(config-if)#ip address 10.1.230.1 255.255.255.0

BR2(config-if)#description Voice

BR2(config-if)#exit

BR2(config)#interface range FastEthernet 0/1/0 - 1

BR2(config-if-range)#switchport voice vlan 230

BR2(config-if-range)#switchport access vlan 220

BR2(config-if-range)#no shut

! Tip - Don’t forget to add the new networks into the routing protocol.

BR2(config)#router ospf 1

BR2(config-router)#network 10.1.220.0 0.0.0.255 area 0

BR2(config-router)#network 10.1.230.0 0.0.0.255 area 0

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2. Validate your configurations with the “show

vlan” ( or “show current” under the vlan database)

and “show cdp neighbors” commands.

!

HQ-3750#show vlan

VLAN Name Status Ports

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

1 default active ! Fa1/0/1, Fa1/0/2, Fa1/0/3, Fa1/0/4,

! ! ! ! ! ! Fa1/0/5, Fa1/0/8, Fa1/0/9, Fa1/0/10,

! ! ! ! ! ! Fa1/0/11, Fa1/0/12, Fa1/0/13, Fa1/0/14,

! ! ! ! ! ! Fa1/0/15, Fa1/0/16, Fa1/0/17, Fa1/0/18,

! ! ! ! ! ! Fa1/0/19, Fa1/0/20, Fa1/0/21, Fa1/0/22,

! ! ! ! ! ! Fa1/0/23, Gi1/0/1, Gi1/0/2

20 Data active

30 Voice active

1002 fddi-default act/unsup

1003 token-ring-default act/unsup

1004 fddinet-default act/unsup

1005 trnet-default act/unsup

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HQ-3750#show cdp neigh

Capability Codes: R - Router, T - Trans Bridge, B - Source Route Bridge

S - Switch, H - Host, I - IGMP, r - Repeater, P - Phone

Device ID Local Intrfce Holdtme Capability Platform Port ID

CCMSUB Fas 1/0/1 152 H none foundeth0

CCMPUB Fas 1/0/1 157 H none foundeth0

HQ_Rtr Fas 1/0/24 140 R S I 2811 Fas 0/0.1

uccx. Fas 1/0/5 172 H Win2000 SeEth 1/1

presence Fas 1/0/4 145 H none foundeth0

SEP003094C27864 Fas 1/0/6 138 H P IP Phone 7Port 1

SEP003094C2DA38 Fas 1/0/7 132 H P IP Phone 7Port 1

SEP0002FDAEFDDF Fas 1/0/8 159 H P IP Phone 7Port 1

Unity Fas 1/0/3 162 H none foundeth0

PSTN-WAN Fas 1/0/22 124 R S I 2811 Fas 0/0

HQ-3750#

!Note - The publisher and subscriber are on the same physical box so they will

show as if they are connected to the same port on the switch. This is normal

and you should configure FastEthernet 1/0/2 as if the subscriber is plugged

into that port.

BR1#vlan database

BR1(vlan)#show current

VLAN ISL Id: 1

Name: default

Media Type: Ethernet

VLAN 802.10 Id: 100001

State: Operational

MTU: 1500

Translational Bridged VLAN: 1002

Translational Bridged VLAN: 1003

VLAN ISL Id: 120

Name: Data

Media Type: Ethernet

VLAN 802.10 Id: 100220

State: Operational

MTU: 1500

VLAN ISL Id: 130

Name: Voice

Media Type: Ethernet

VLAN 802.10 Id: 100230

State: Operational

MTU: 1500

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BR1#show cdp neigh

Capability Codes: R - Router, T - Trans Bridge, B - Source Route Bridge

S - Switch, H - Host, I - IGMP, r - Repeater

Device ID Local Intrfce Holdtme Capability Platform Port ID

HQ_Rtr Ser 0/1/0.1 161 R S I 2811 Ser 0/2/0.1

SEP0016C72D3DCC Fas 0/3/1 146 H IP Phone Port 1

SEP003094C3DCF9 Fas 0/3/0 143 H IP Phone Port 1

BR1#

BR2# show cdp neigh

Capability Codes: R - Router, T - Trans Bridge, B - Source Route Bridge

S - Switch, H - Host, I - IGMP, r - Repeater

Device ID Local Intrfce Holdtme Capability Platform Port ID

HQ_Rtr Ser 0/2/0.1 133 R S I 2811 Ser 0/2/0.2

SEP001F6C7FD6C1 Fas 0/1/1 167 H IP Phone Port 1

SEP001F6C7F722B Fas 0/1/0 167 H IP Phone Port 1

BR2#

BR2#vlan database

BR2(vlan)#show current

VLAN ISL Id: 1

Name: default

Media Type: Ethernet

VLAN 802.10 Id: 100001

State: Operational

MTU: 1500

Translational Bridged VLAN: 1002

Translational Bridged VLAN: 1003

VLAN ISL Id: 220

Name: Data

Media Type: Ethernet

VLAN 802.10 Id: 100220

State: Operational

MTU: 1500

VLAN ISL Id: 230

Name: Voice

Media Type: Ethernet

VLAN 802.10 Id: 100230

State: Operational

MTU: 1500

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3. In order to accomplish this we need to first create

the dhcp excluded address and then define the DHCP

pool:

HQ_Rtr(config)#ip dhcp excluded-address 10.1.30.1 10.1.30.119

HQ_Rtr(config)#ip dhcp excluded-address 10.1.30.131 10.1.30.255

HQ_Rtr(config)#ip dhcp pool IPPHONE

HQ_Rtr(config-config)#network 10.1.30.0 255.255.255.0

HQ_Rtr(config-config)#default-router 10.1.30.1

HQ_Rtr(config-config)#option 150 ip 10.1.200.21 10.1.200.25

Time Saver - In order to save time you should use notepad to copy and

paste configurations which may be the same or need little modification.

4. We can use the same configuration from the last task

but modify for BR1:

BR1(config)#ip dhcp excluded-address 10.1.130.1 10.1.130.129

BR1(config)#ip dhcp excluded-address 10.1.130.141 10.1.130.255

BR1(config)#ip dhcp pool IPPHONE

BR1(config-config)#network 10.1.130.0 255.255.255.0

BR1(config-config)#default-router 10.1.130.1

BR1(config-config)#option 150 ip 10.1.200.21 10.1.200.25

5. Perform the same for BR2:

BR2(config)#ip dhcp excluded-address 10.1.230.1 10.1.230.209

BR2(config)#ip dhcp excluded-address 10.1.230.221 10.1.230.255

BR2(config)#ip dhcp pool IPPHONE

BR2(config-config)#network 10.1.230.0 255.255.255.0

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BR2(config-config)#default-router 10.1.230.1

BR2(config-config)#option 150 ip 10.1.230.1

6. Next configure each router for the appropriate time

zone and NTP settings:

HQ_Rtr(config)#clock timezone PST -8

HQ_Rtr(config)#clock summer-time PDT recurring

HQ_Rtr(config)#ntp master 2

BR1(config)#clock timezone EST -5

BR1(config)#clock summer-time EDT recurring

BR1(config)#ntp server 10.10.32.1

BR2(config)#clock timezone UTC +5 30

BR2(config)#clock summer-time PDT recurring

BR2(config)#ntp server 10.10.32.1

! Tip - Use can use your candidate PC to look up the time zone of most

countries but most likely they will specify for you.

! Tip - Don’t forget to set the time zone settings in CallManager as it may not be

mentioned in the phone setup tasks.

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CallManager and CallManager Express:

1. This tasks requires a lot of configuration as this is

the base line for everything you do in CallManager. We

will setup everything from device pools, locations,

regions, media resource lists and groups, route lists

and route groups, partitions and calling search

spaces.

!

Lets start off with service activation first as

nothing will work if we don’t have services activated:

Make sure you enable services on both servers and all

services except the “Cisco Messaging Interface”.

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!Tip - Enabling all services as stated above will ensure that all needed services

are activated. Unless you are specifically told to only enable needed services

then this strategy will allow you to make sure all the required services are

running.

Next we need to start configuring the required

CallManager settings needed to allow for the

registration of HQ and BR1 phones.

Make sure the proper servers are defined:

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Next create a CallManager group that will allow for

devices to register to the subscriber and then the

publisher:

When finished you should see the following to groups:

! Note - The requirement for registration order is normally on the first page of

instructions.

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Next we need to create the regions:

We need to make sure the settings between the regions

are set appropriately:

Time Saver - This is where reading through the whole lab will save a lot of

time as we can set the proper region settings and never return to this page.

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Now we need to create the HQ and BR1 Date/Time groups:

! Note - This is also where we could set the date/time format if asked during a

specific task.

Next we need to create the locations:

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Time Saver - This is where reading through the whole lab will save a lot of

time as we can set the proper location settings and never return to this page.

We now need to create the media resource groups and

lists:

!Tip - In reading the lab we know that there is a need to separate the MOH

servers and it is always a good idea to separate hardware from software

resources.

We can then create the needed media resource lists:

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Below is the full list of the media resource lists:

We now need to create the device pools:

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Time Saver - We know from reading the lab that one phone will not be

allowed to register in SRST. We can create the device pool for this function

now in order to save time later.

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Here is the final list of device pools:

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Next we need to check the Enterprise Parameters and

make changes as needed:

!Tip - Changing the host name to an IP address in the URL settings will prevent

authentication and IP phone service failures. This is highly recommended

whether you have a DNS server or not!

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We can also create the AAR groups now as this will

help to cut down the number of times we need to visit

the phone configuration page:

Once we have both groups created we need to set the

dial prefix according to the dial plan:

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Now we will need to create the the partitions and

calling search spaces that will be needed for the lab:

On the next page is a table of calling search spaces

and the partitions that belong to each calling search

space.

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!"##$%&'()"*+,'(-"+)'

."/)' 0)1+*$-2$3%' 4352)'6"*2$2$3%1'73*'2,$1'!"##$%&'()"*+,'(-"+)'

!""#$%&#''%( !""#$%&#''%( )*+$%&+''%(

!""#$%&#,-*( !""#$%&#,-*( )*+$%&+.&&(

)*+$%&+,/012/34(

)*+$%&+*566$7)'""(

)*+89+,/012/34(

)*+$%:+,/012/34(

)*+$%&+6;<34(

)*+$%&+6=(

)*+$%&+,-*(

!""#$%&#,-*>%-'6( !""#$%&#,-*>%-'6( )*+$%&+.&&(

)*+$%&+,/012/34(

)*+$%&+*566$7)'""(

)*+89+,/012/34(

)*+$%:+,/012/34(

!""#$%&#6=( !""#$%&#6=( )*+$%&+.&&(

)*+$%&+,/012/34(

)*+$%&+*566$7)'""(

)*+89+,/012/34(

)*+$%:+,/012/34(

)*+$%&+6;<34(

)*+$%&+6=(

!""#$%&#65!'6( !""#$%&#65!'6( )*+$%&+.&&(

)*+$%&+,/012/34(

)*+$%&+*566$7)'""(

)*+89+,/012/34(

)*+$%:+,/012/34(

)*+$%&+6;<34(

!""#$%:( !""#$%:( )*+$%&+,/012/34(

)*+$%:+,/012/34(

)*+89+,/012/34(

)*+$%:+*566$7)'""(

!""#89#''%( !""#89#''%( )*+89+''%(

!""#89#,-*( !""#89#,-*( )*+89+.&&(

)*+89+,/012/34(

)*+89+*566$7)'""(

)*+$%&+,/012/34(

)*+$%:+,/012/34(

)*+89+6;<34(

)*+89+6=(

)*+89+,-*(

!""#89#,-*>%-'6( !""#89#,-*>%-'6( )*+89+.&&(

)*+89+,/012/34(

)*+89+*566$7)'""(

)*+$%&+,/012/34(

)*+$%:+,/012/34(

!""#89#6=( !""#89#6=( )*+89+.&&(

)*+89+,/012/34(

)*+89+*566$7)'""(

)*+$%&+,/012/34(

)*+$%:+,/012/34(

)*+89+6;<34(

)*+89+6=(

!""#89#65!'6( !""#89#65!'6( )*+89+.&&(

)*+89+,/012/34(

)*+89+*566$7)'""(

)*+$%&+,/012/34(

)*+$%:+,/012/34(

)*+89+6;<34(

!""#?5,!>@',6( !""#?5,!>@',6( )*+$%&+,/012/34(

)*+89+,/012/34(

(

(

We should now be able to add the phones per the

requirements in the table.

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HQ Phone 1:

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HQ Phone 2:

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BR1 Phone 1:

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BR1 Phone 2:

!Note - In reading the lab we know the BR1 Phone 2 requires another extension

“*2002”. We can create that as part of the initial phone setup which we have

done here or you could wait till that specific task.

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BR1 Phone 2 (Line 2):

! Tip - Most of the phone settings per phone are the same per site except for the

calling search space.

Time Saver - In order to save time you can use auto registration on the

callmanager and then modify each phone to the correct settings.

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We can now start the registration of phones within the

BR2 site. We will first need to define all the

required settings underneath the “telephony-service”

and “voice register global” configuration. We can then

create the phones and their associated DN’s:

BR2(config)#telephony-service

BR2(config-telephony)#authentication credential admin cisco

BR2(config-telephony)#max-ephones 2

BR2(config-telephony)#max-dn 5

BR2(config-telephony)#ip source-address 10.1.230.1 port 2000

BR2(config-telephony)#voicemail 3180

BR2(config-telephony)#max-conferences 8 gain -6

BR2(config-telephony)#call-park system application

BR2(config-telephony)#call-forward pattern .T

BR2(config-telephony)#dn-webedit

BR2(config-telephony)#time-webedit

BR2(config-telephony)#transfer-system full-blind

BR2(config-telephony)#transfer-pattern 9.T

BR2(config-telephony)#fac standard

BR2(config-telephony)#create cnf-files

BR2(config-telephony)#exit

BR2(config)#voice register global

BR2(config-register-global)#mode cme

BR2(config-register-global)#source-address 10.1.230.1 port 5060

BR2(config-register-global)#max-dn 10

BR2(config-register-global)#max-pool 2

BR2(config-register-global)#load 7941 SIP41.8-3-3S

BR2(config-register-global)#mwi stutter

BR2(config-register-global)#mwi reg-e164

BR2(config-register-global)#voicemail 3180

BR2(config-register-global)#tftp-path flash:

BR2(config-register-global)#create profile

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Now we can configure the phones and their DN’s:

BR2(config)#voice register dn 1

BR2(config-register-dn)#number 3001

BR2(config-register-dn)#call-forward b2bua noan 3180 timeout 20

BR2(config-register-dn)#pickup-call any-group

BR2(config-register-dn)#pickup-group 1

BR2(config-register-dn)#name BR2 Phone 1

BR2(config-register-dn)#mwi

BR2(config-register-dn)#voice register dn 2

BR2(config-register-dn)#number 3002

BR2(config-register-dn)#call-forward b2bua noan 3180 timeout 20

BR2(config-register-dn)#pickup-call any-group

BR2(config-register-dn)#pickup-group 1

BR2(config-register-dn)#name BR2 Phone 2

BR2(config-register-dn)#mwi

We can also configure the extra DN specified for the

shared line on BR2 phone 1 and 2:

BR2(config-register-dn)#voice register dn 3

BR2(config-register-dn)#number 3003

BR2(config-register-dn)#shared-line max-calls 6

We can now configure the phones:

BR2(config)#voice register pool 1

BR2(config-register-pool)#id mac 001F.6C7F.D6C1

BR2(config-register-pool)#type 7941

BR2(config-register-pool)#number 1 dn 1

BR2(config-register-pool)#number 2 dn 3

BR2(config-register-pool)#dtmf-relay rtp-nte

BR2(config-register-pool)#codec g711ulaw

BR2(config-register-pool)#no keep-conference

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BR2(config-register-pool)#voice register pool 2

BR2(config-register-pool)#id mac 001F.6C7F.722B

BR2(config-register-pool)#type 7941

BR2(config-register-pool)#number 1 dn 2

BR2(config-register-pool)#number 2 dn 3

BR2(config-register-pool)#dtmf-relay rtp-nte

BR2(config-register-pool)#codec g711ulaw

BR2(config-register-pool)#no keep-conference

Next we need to allow the SIP firmware to be

downloaded from the router flash:

BR2(config)# tftp-server flash:SIP41.8-4-4S.loads

BR2(config)# tftp-server flash:term41.default.loads

BR2(config)# tftp-server flash:SIP41.8-3-3S.loads

BR2(config)# tftp-server flash:jar41sip.8-3-2-27.sbn

BR2(config)# tftp-server flash:dsp41.8-3-2-27.sbn

BR2(config)# tftp-server flash:cvm41sip.8-3-2-27.sbn

BR2(config)# tftp-server flash:cnu41.8-3-2-27.sbn

BR2(config)# tftp-server flash:apps41.8-3-2-27.sbn

!Tip - If you like to use the CME and CUE web interface the the following

commands should be entered at this stage to prepare you to use the web

interface.

BR2(config)#ip http server

BR2(config)#no ip http secure-server

BR2(config)#ip http path flash:

BR2(config)#telephony-service

BR2(config-telephony)#web admin system name admin password cisco

BR2(config-telephony)#url services http://10.1.230.2/voiceview/common/

login.do (Inserted by default in most cases)

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BR2(config-telephony)#url authentication http://10.1.230.1/CCMCIP/

authenticate.asp (Inserted by default in most cases)

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2. This was already completed in the above solution with the

“voice register dn 3” configuration and the addition of

this button to each “voice register pool”.

3. This was already completed in the above solution with the

“no keep-conference” command under the “voice register

pool” configuration.

4. This was accomplished in task 1 by setting the alerting

name, caller id, external phone number mask, and date/

time group format the same in CallManager.

5. This was accomplished in task 1 by setting setting the

“pickup-group 1” command under all the DN’s.

6. This is accomplished with the following configuration:

BR2(config)#voice hunt-group 1 sequential

BR2(config-voice-hunt-group)#final 3000

BR2(config-voice-hunt-group)#list 3001,3002

BR2(config-voice-hunt-group)#timeout 10

BR2(config-voice-hunt-group)#pilot 3111

!Note - Though the task asks us to allow the number to be reach by the full

E164 number we can perform this action later when setting up the call routing

as all calls will need to be striped to four digits.

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Voice Gateways and Signaling

1. In order to enable the HQ router as an MGCP gateway we

need the following configuration on the HQ router:

HQ_Rtr(config)#network-clock-participate wic 0

HQ_Rtr(config)#isdn switch-type primary-ni

HQ_Rtr(config)#voice-card 0

HQ_Rtr(config-voicecard)#dspfarm

HQ_Rtr(config-voicecard)#dsp services dspfarm

HQ_Rtr(config-voicecard)#exit

HQ_Rtr(config)#controller T1 0/0/0

HQ_Rtr(config-controller)#cablelength long 0db

HQ_Rtr(config-controller)#pri-group timeslots 1-3,24 service mgcp

HQ_Rtr(config-controller)#exit

HQ_Rtr(config)#interface Serial0/0/0:23

HQ_Rtr(config-if)# isdn bind-l3 ccm-manager

HQ_Rtr(config-if)#exit

HQ_Rtr(config)#ccm-manager redundant-host 10.1.200.21

HQ_Rtr(config)#ccm-manager mgcp

HQ_Rtr(config)#mgcp

HQ_Rtr(config)#mgcp call-agent 10.1.200.25 service-type mgcp version 0.1

HQ_Rtr(config)#mgcp fax t38 ecm

HQ_Rtr(config)#mgcp

! Note - In IOS release 12.4 the dial-peer no longer needs to be created with the

service command.

Time Saver - If you have multiple MGCP or H.323 gateways then use

notepad to copy and paste common configuration components.

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Next we will need to configure the gateway in

CallManager:

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!Tip - If you have set or been asked to set a domain name on the router the

device name for MGCP would be the router name plus the domain name. So in

our case it could be “HQ_Rtr.voiceie.com” if we had “voiceie.com” as a

domain name on the router.

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2. In order to enable the BR1 router as an H.323 gateway we

need the following configuration on the BR1 router:

BR1(config)#network-clock-participate wic 1

BR1(config)#isdn switch-type primary-ni

BR1(config)#voice-card 0

BR1(config-voicecard)#dsp services dspfarm

BR1(config-voicecard)#exit

BR1(config)#voice translation-rule 1

BR1(cfg-translation-rule)#rule 1 /\(^702333\)\(2...\)/ /\2/

BR1(cfg-translation-rule)#exit

BR1(config)#voice translation-profile STRIP

BR1(cfg-translation-profile)#translate called 1

BR1(cfg-translation-profile)#exit

BR1(config)#voice-port 0/1/0:23

BR1(config-voiceport)#translation-profile incoming STRIP

BR1(config-voiceport)#exit

BR1(config)#controller T1 0/1/0

BR1(config-controller)#framing esf

BR1(config-controller)# linecode b8zs

BR1(config-controller)# pri-group timeslots 1-3,24

BR1(config-controller)# exit

BR1(config)#interface Loopback0

BR1(config-if)#h323-gateway voip interface

BR1(config-if)#h323-gateway voip bind srcaddr 10.10.32.2

BR1(config-if)#exit

BR1(config)#interface Serial0/1/0:23

BR1(config-if)#isdn outgoing display-ie

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Next we will need to setup BR1 as an H.323 gateway in

CallManager:

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!Tip - The most common issue with H.323 gateways is that IP address specified

in CallManager is not the IP address that the H.323 process in the router is

binding. Make sure these IP address match!

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3. In order to enable the BR2 router as an H.323 gateway we

need the following configuration on the BR2 router:

BR2(config)#network-clock-participate wic 0

BR2(config)#isdn switch-type primary-net5

BR2(config)#voice translation-rule 1

BR2(cfg-translation-rule)#rule 1 /\(^44207630\)\(3...\)/ /\2/

BR2(cfg-translation-rule)#exit

BR2(config)#voice translation-profile STRIP

BR2(cfg-translation-profile)#translate called 1

BR2(cfg-translation-profile)#exit

BR2(config)#voice-card 0

BR2(config-voicecard)#dspfarm

BR2(config-voicecard)#dsp services dspfarm

BR2(config-voicecard)#exit

BR2(config)#controller E1 0/0/0

BR2(config-controller)#pri-group timeslots 1-3,16

BR2(config-controller)#exit

BR2(config)#interface Serial0/0/0:15

BR2(config-if)#isdn outgoing display-ie

BR2(config-if)#exit

BR2(config)#voice-port 0/0/0:15

BR2(config-voiceport)#translation-profile incoming STRIP

4. In order to setup the gatekeeper we need to perform the

following configuration on the HQ router:

HQ_Rtr(config)#gatekeeper

HQ_Rtr(config-gk)#zone local CM ccbootcamp.com 10.10.32.1

HQ_Rtr(config-gk)#zone local CME ccbootcamp.com

HQ_Rtr(config-gk)#gw-type-prefix 1#* default-technology

HQ_Rtr(config-gk)#no shutdown

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Next we need to enable CallManager to register to the

gatekeeper through a H.225 gatekeeper controlled

trunk. We can accomplish this by adding the gatekeeper

and then the trunk in CallManager:

!Tip - If you would like to lower the time it takes for gatekeeper to register

after a loss of connectivity then change the “Registration Retry Timeout” to 30

seconds.

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Now we need to add the trunk:

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! Tip - You can validate registration with the “show gatekeeper endpoints”

command on the HQ router.

HQ_Rtr#show gatekeeper endpoints

GATEKEEPER ENDPOINT REGISTRATION

================================

CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags

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

10.1.200.21 33072 10.1.200.21 32794 CM H323-GW

H323-ID: BR2_1

Voice Capacity Max.= Avail.= Current.= 0

10.1.200.25 33077 10.1.200.25 32785 CM H323-GW

H323-ID: BR2_2

Voice Capacity Max.= Avail.= Current.= 0

Total number of active registrations = 4

HQ_Rtr#

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Next we will need to configure the BR2 router is

register with the gatekeeper. This requires the

following configuration:

BR2(config)#interface Loopback0

BR2(config-if)#h323-gateway voip interface

BR2(config-if)#h323-gateway voip id CME ipaddr 10.10.32.1 1719

BR2(config-if)#h323-gateway voip h323-id CME

BR2(config-if)#h323-gateway voip tech-prefix 1#

BR2(config-if)#h323-gateway voip bind srcaddr 10.10.32.3

BR2(config-if)#exit

BR2(config)#gateway

! Tip - Once again you can verify with the “show gatekeeper endpoints”

command on the HQ router.

HQ_Rtr#show gatekeeper endpoints

GATEKEEPER ENDPOINT REGISTRATION

================================

CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags

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

10.10.32.3 1720 10.10.32.3 57205 CM H323-GW

H323-ID: CME

Voice Capacity Max.= Avail.= Current.= 0

10.1.200.21 33072 10.1.200.21 32794 CM H323-GW

H323-ID: BR2_1

Voice Capacity Max.= Avail.= Current.= 0

10.1.200.25 33077 10.1.200.25 32785 CM H323-GW

H323-ID: BR2_2

Voice Capacity Max.= Avail.= Current.= 0

Total number of active registrations = 4

HQ_Rtr#

!Note - You will notice that the directory numbers from the SIP CME did not

register with gatekeeper. This is to be expected as SIP CME phones cannot be

auto registered to gatekeeper.

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Call Routing

1. The has already been accomplished for HQ and BR1 when

we setup the phones previously by giving them the

proper calling search space. To enable class of

service for the BR2 site we need to create the

following COR list:

dial-peer cor custom

name Local

name LD

name INT

name Emergency

dial-peer cor list Emergency

member Emergency

dial-peer cor list Local

member Local

dial-peer cor list Int

member INT

dial-peer cor list LD

member LD

dial-peer cor list Phone1

member Local

member LD

member INT

member Emergency

dial-peer cor list Phone2

member Local

member LD

member Emergency

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Now that we have created the COR list we need to

assign the proper COR list to the voice register pool

of the proper phone:

BR2(config)#voice register pool 1

BR2(config-register-pool)#cor incoming Phone1 default

BR2(config-register-pool)#voice register pool 2

BR2(config-register-pool)#cor incoming Phone2 default

2. In order to accomplish the next two tasks we need to

create all the route groups and route lists first. We

will first need to create the route groups:

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Then next the route lists in the following tables with

the proper settings:

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(%+D2"J'(%)%*+' !"#(/*'

1233#J'!2"*L'M"20+$/"6'52+O' '

!"#$%&'(%)%*+',-.*)/%0)'1233+4' '

5#67#"+' 89:8;9:(8<=9>?@AB*"',C33'!/"*+4'

'

9!:6-8&

1233%0)'!2"*L'M"20+$/"62*%/0+'

;+#'1233%0)'!2"*LN+'I&*#"023'52+O' -0'

1233%0)'!2"*L'M"20+$/"6'52+O' '

!"#$%&'(%)%*+',-.*)/%0)'1233+4' '

1233#J'!2"*L'M"20+$/"62*%/0+'

(%+D2"J'(%)%*+' !"#(/*'

1233#J'!2"*L'M"20+$/"6'52+O' '

!"#$%&'(%)%*+',-.*)/%0)'1233+4' P'

5#67#"+' <9Q<9QRSQS',C33'!/"*+4'

'''

456';<='&

(#+D"%E*%/0' ?@AF-1CF'

1;15'G"/.E' 8;H=!;H'

I0273#J' K'

'

456-8&

1233%0)'!2"*L'M"20+$/"62*%/0+'

;+#'1233%0)'!2"*LN+'I&*#"023'52+O' -0'

1233%0)'!2"*L'M"20+$/"6'52+O' '

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!"#$%&'()$&

!"#$%&'()$& *))"+(*$%,&!"#$%&-."#/&0($1&2%23%.)&

! "#$%&'!(&)&*+!,-.*)/&0)!1233+4! !

1233$5!"2#*6!7#20+%/#82*&/0+!

(&+92#5!(&)&*+! "#$(/*!

1233$5!"2#*6!7#20+%/#8!:2+;! !

"#$%&'!(&)&*+!,-.*)/&0)!1233+4! !

:$8<$#+! =>?=@>?(=AB>CDEFG*#!,H33!"/#*+4!

!

4!56-7&

1233&0)!"2#*6!7#20+%/#82*&/0+!

@+$!1233&0)!"2#*6I+!J'*$#023!:2+;! -0!

1233&0)!"2#*6!7#20+%/#8!:2+;! !

"#$%&'!(&)&*+!,-.*)/&0)!1233+4! !

1233$5!"2#*6!7#20+%/#82*&/0+!

(&+92#5!(&)&*+! "#$(/*!

1233$5!"2#*6!7#20+%/#8!:2+;! !

"#$%&'!(&)&*+!,-.*)/&0)!1233+4! KAL>M!

:$8<$#+! A>NA>NOMNM!,H33!"/#*+4!

!!!

!

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A summary of all the route lists is below:

We can now create all the route patterns for the HQ

call routing and point them to the proper route list:

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! Note - All the digit manipulation and phone number masking is done in the

route lists and therefore should not be done in the route pattern.

Next we need to add the proper call routing to the

gatekeeper on the HQ gateway:

HQ_Rtr(config-gk)#zone prefix CME 3...

HQ_Rtr(config-gk)#zone prefix CM 7....

HQ_Rtr(config-gk)#zone prefix CME 96......

3. We can now use the previous created route groups and

route lists to setup the BR1 call routing by adding

the following route patterns:

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Now since the BR1 gateway is an H.323 gateway we need

to add the proper dial-peers to the BR1 gateway to

complete the call routing. We may also want to take

into account the SRST task for maintaining class of

service and create the COR list at the same time:

dial-peer cor custom

name Local

name LD

name INT

name Emergency

dial-peer cor list Emergency

member Emergency

dial-peer cor list Local

member Local

dial-peer cor list Int

member INT

dial-peer cor list LD

member LD

dial-peer cor list Phone1

member Local

member LD

member INT

member Emergency

dial-peer cor list Phone2

member Local

member LD

member Emergency

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We can now create the dial-peers with the COR already

configured:

dial-peer voice 1 pots

incoming called-number .

direct-inward-dial

dial-peer voice 2000 voip

preference 1

destination-pattern 2...

session target ipv4:10.1.200.25

ip qos dscp cs3 signaling

dial-peer voice 2001 voip

preference 2

destination-pattern 2...

session target ipv4:10.1.200.21

ip qos dscp cs3 signaling

dial-peer voice 7 pots

corlist outgoing Local

destination-pattern 9[2-9]......

port 0/1/0:23

forward-digits 7

dial-peer voice 11 pots

corlist outgoing LD

destination-pattern 91[2-9]..[2-9]......

port 0/1/0:23

forward-digits 11

!

dial-peer voice 911 pots

corlist outgoing Emergency

destination-pattern 911

port 0/1/0:23

forward-digits all

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!

dial-peer voice 9911 pots

corlist outgoing Emergency

destination-pattern 9911

port 0/1/0:23

forward-digits 3

!

dial-peer voice 9011 pots

corlist outgoing Int

destination-pattern 9011T

port 0/1/0:23

prefix 011

4. We can now setup the call routing for BR2 with the

following configuration on the BR2 router. We first

need to allow all the communications that maybe needed

in our environment:

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

supplementary-service h450.12

no supplementary-service sip refer

fax protocol cisco

h323

sip

registrar server expires max 1200 min 300

Next we need to create a voice class codec for our

calls through gatekeeper so that the call setup will

not fail:

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voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g729r8

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Now we can create all the needed dial-peers with the

appropriate COR list which were previously defined:

dial-peer voice 1 pots

incoming called-number .

direct-inward-dial

!

dial-peer voice 2 pots

corlist outgoing Local

destination-pattern 90[1-9].......

port 0/0/0:15

prefix 0

!

dial-peer voice 1000 voip

preference 1

destination-pattern 71...

voice-class codec 1

session target ras

!

dial-peer voice 1001 pots

preference 2

destination-pattern 71...

port 0/0/0:15

prefix 0017029461

!

dial-peer voice 2000 voip

preference 1

destination-pattern 72...

voice-class codec 1

session target ras

!

dial-peer voice 2001 pots

preference 2

destination-pattern 72...

port 0/0/0:15

prefix 0017023332

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!

dial-peer voice 7 pots

corlist outgoing Local

destination-pattern 9[1-9]......

port 0/0/0:15

forward-digits 7

!

dial-peer voice 999 pots

corlist outgoing Emergency

destination-pattern 999

port 0/0/0:15

forward-digits all

!

dial-peer voice 900 pots

corlist outgoing Int

destination-pattern 900.T

port 0/0/0:15

prefix 00

We also need to add a translation pattern to strip the

“7” as the call comes into callmanager from BR2:

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We will also need to enable transcoding on the BR2

gateway to support calls transgressing from HQ/BR1 to

BR2:

sccp local FastEthernet0/0.230

sccp ccm 10.1.230.1 identifier 1 priority 1 version 3.1

sccp

sccp ccm group 1

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associate ccm 1 priority 1

associate profile 1 register TRANSCODE

dspfarm profile 1 transcode

codec g711ulaw

codec g711alaw

codec g729r8

codec g729abr8

codec g729ar8

codec g729br8

maximum sessions 4

associate application SCCP

telephony-service

sdspfarm units 2

sdspfarm transcode sessions 4

sdspfarm tag 1 TRANSCODE

!Tip - At this point you can start testing your calls to make sure they all

complete correctly. In order to test redundancy you can shutdown the

gatekeeper or the corresponding T1/E1 controller to test.

Time Saver - By saving the testing until your call routing is complete you

can save time by not interrupting your configuration and then fixing as you

test.

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Codec and Call Admission Control

1. This task was already accomplished earlier when we

created the regions.

2. This task was already accomplished earlier when we

created the regions.

3. This task was completed when we create the locations

earlier.

4. In order to accomplish this task we need to enable

zone bandwidth control on the gatekeeper:

HQ_Rtr(config)#gatekeeper

HQ_Rtr(config-gk)#bandwidth total zone CME 48

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High Availability Features

1. In order to accomplish this task we need to enable

“call-manager-fallback” in the BR1 gateway:

application

global

service alternate DEFAULT

call-manager-fallback

secondary-dialtone 9

max-conferences 4 gain -6

ip source-address 10.1.130.1 port 2000

max-ephones 2

max-dn 10

transfer-pattern .T

voicemail 917029461170

call-forward pattern .T

2. This task was accomplished earlier by setting the BR1

phone 1 device pool to “BR1-NOSRST” which had the SRST

reference disabled.

3. This was completed during the call routing section as

the dial-peers were created with 9 at the beginning.

4. In order to accomplish this we need to add the proper

COR list to the DN of BR1 phone 2 in the incoming

direction:

call-manager-fallback

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cor incoming Phone2 1 2002

5. Most of this task has already been completed during

the phone setup section. We created the AAR groups

with the appropriate prefixes and then assigned the

DN’s to the appropriate AAR group. The last step is to

enable the AAR service via the callmanager service

parameters:

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Media Resources Management

1. In order to accomplish this task we first need to

create the resource in callmanager:

Next we need to configure the SCCP and DSPFARM

settings on the HQ router:

sccp local Loopback0

sccp ccm 10.1.200.21 identifier 2 priority 2 version 7.0

sccp ccm 10.1.200.25 identifier 1 priority 1 version 7.0

sccp

!

sccp ccm group 1

associate ccm 1 priority 1

associate ccm 2 priority 2

associate profile 2 register HQTRANS

associate profile 1 register HQCONF

!

dspfarm profile 2 transcode

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codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

maximum sessions 4

associate application SCCP

!

dspfarm profile 1 conference

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

codec g729r8

codec g729br8

maximum sessions 4

associate application SCCP

!Tip - You can use the “show sccp” and the show “dspfarm profile” command

to verify gateway configuration. You may need to disable and then enable the

SCCP function if the resources aren’t registering.

2. In order to accomplish this task we first need to

create the resource in callmanager:

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Time Saver - This is another great instance of where copy and paste can

help you as the configuration will be very similar to the HQ router.

sccp local Loopback0

sccp ccm 10.1.200.25 identifier 1 priority 1 version 7.0

sccp ccm 10.1.200.21 identifier 2 priority 2 version 7.0

sccp

!

sccp ccm group 1

associate ccm 1 priority 1

associate ccm 2 priority 2

associate profile 1 register BR1CONF

associate profile 2 register BR1TRANS

!

dspfarm profile 2 transcode

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

codec gsmfr

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maximum sessions 4

associate application SCCP

!

dspfarm profile 1 conference

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

codec g729r8

codec g729br8

maximum sessions 4

associate application SCCP

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3. To accomplish this task we first need to create the

resource in callmanager:

The gateway configuration portion was accomplished

above in the conference bridge section.

4. To accomplish this task we first need to create the

resource in callmanager:

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The gateway configuration portion was accomplished

above in the conference bridge section.

5. In a previous step we create all the media resource

groups and list. Now we can add the proper media

resources to the proper groups. The publisher MOH

server will need to be put into the “PUB_MOH” media

resource group and we will need to make sure that this

is the MOH only server listed in the media resource

list for HQ.

6. We need to perform the same actions for BR1 but with

the subscriber server being put into the “SUB_MOH”

media resource group and making sure that this is the

only MOH server listed in the media resource list for

BR1. In addition we will need to enable the default

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audio source, subscriber MOH server, and the “SUB_MOH”

media resource group for multicasting:

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We need to insure that the “Max Hops” is set to “1” so

that the multicast stream stays local.

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Now we need to enable the G.729 codec through the

Cisco IP Voice Media Streaming App service parameters:

We then need to configure the BR1 gateway to support

multicast and the router to play the MOH file from

flash:

call-manager-fallback

moh music-on-hold.au

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multicast moh 239.1.1.1 port 16388 route 10.1.130.1

ip multicast-routing

interface Vlan130

ip pim sparse-dense-mode

! Tip - To test place a call from HQ to BR1 and then put the call on hold from

both ends. You should receive MOH but hear different music on both sides.

7. In order to accomplish this we first need to create

the MeetMe conference resources in callmanager:

We then need to change the callmanager service

parameter to allow for 10 participants per conference:

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8. This is accomplished through the same callmanager

service parameter section as the task above by setting

the “Drop Ad Hoc Conference” parameter correctly.

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QoS Features

1. We can accomplish this with the following

configuration.

On the HQ gateway:

interface Serial0/2/0.1 point-to-point

bandwidth 384

frame-relay interface-dlci 201

auto qos voip trust fr-atm

On the BR1 Gateway:

interface Serial0/2/0.1 point-to-point

bandwidth 384

frame-relay interface-dlci 101 ppp

auto qos voip trust fr-atm

The on both gateways modify the auto generated policy

map per the requirements:

policy-map AutoQoS-Policy-Trust

class AutoQoS-VoIP-RTP-Trust

priority percent 33

class AutoQoS-VoIP-Control-Trust

bandwidth percent 5

class class-default

fair-queue

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2. We can accomplish this with the following

configuration.

On the HQ gateway:

interface Serial0/2/0.2 point-to-point

bandwidth 1200

frame-relay interface-dlci 202

auto qos voip trust

On the BR2 gateway:

interface Serial0/2/0.1 point-to-point

bandwidth 1200

frame-relay interface-dlci 102

auto qos voip trust

Then on both gateways modify the auto generated policy

map per the requirements:

policy-map AutoQoS-Policy-Trust2

class AutoQoS-VoIP-RTP-Trust

priority 360

class AutoQoS-VoIP-Control-Trust

bandwidth 60

class class-default

fair-queue

3. This was accomplished earlier under the two dial-peers

pointing traffic back to the callmanagers.

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Application and End Users

In order to move forward past this section we need to

define all the users. Outlined below in tables are the

users that need to be created and the settings for those

users.

!Tip - Pay close attention to the associated devices and rights of each user.

Before you start to add user you will need to create a “Standard AXL User”

group and assign the “Standard AXL API Access” role to that group.

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!""#$%&'$()*+,-.*

+,-.*/0* /)1(.2&'$()*

!"#$%&&

'(()#*+"#,-&./01&2-3,14+"#,-&

510/0-*0&$1,6(& 7"+-8+18&510/0-*0&91,6(&

'**0("&510/0-*0&76:/*1#("#,-& ;&

'**0("&<6"=,3=8#+),9&>?@?>& ;&

'**0("&.-/,)#*#"08&;,"#3#*+"#,-& ;&

'**0("&>0()+*0/&A0+801& ;&

B0C#*0&2-3,14+"#,-&

!,-"1,))08&B0C#*0/& 7?5DDEF!GHBIB!!&7?5DDIDJK!HGLFK&7?5DDIDJK!HB'IL&7?5DDIDJK!IB!@J&

!'5@&2-3,14+"#,-&

'//,*#+"08&!'5@&51,3#)0/& &

5014#//#,-/&2-3,14+"#,-&

$1,6(/& 7"+-8+18&!M2&')),%&!,-"1,)&,3&'))&B0C#*0/&7"+-8+18&!M2&?-+:)08&

>,)0/& 7"+-8+18&!M2&')),%&!,-"1,)&,3&'))&B0C#*0/&7"+-8+18&!M2&?-+:)08&

&&

25!!&&

'(()#*+"#,-&./01&2-3,14+"#,-&

510/0-*0&$1,6(& 7"+-8+18&510/0-*0&91,6(&

'**0("&510/0-*0&76:/*1#("#,-& ;&

'**0("&<6"=,3=8#+),9&>?@?>& ;&

'**0("&.-/,)#*#"08&;,"#3#*+"#,-& ;&

'**0("&>0()+*0/&A0+801& ;&

B0C#*0&2-3,14+"#,-&

!,-"1,))08&B0C#*0/& &

!'5@&2-3,14+"#,-&

'//,*#+"08&!'5@&51,3#)0/& &

5014#//#,-/&2-3,14+"#,-&

$1,6(/& 7"+-8+18&'NO&./01&

>,)0/& 7"+-8+18&'NO&'52&'**0//&

&&

.-#"P&&

'(()#*+"#,-&./01&2-3,14+"#,-&

510/0-*0&$1,6(& 7"+-8+18&510/0-*0&91,6(&

'**0("&510/0-*0&76:/*1#("#,-& ;&

'**0("&<6"=,3=8#+),9&>?@?>& ;&

'**0("&.-/,)#*#"08&;,"#3#*+"#,-& ;&

'**0("&>0()+*0/&A0+801& ;&

B0C#*0&2-3,14+"#,-&

!,-"1,))08&B0C#*0/& &

!'5@&2-3,14+"#,-&

'//,*#+"08&!'5@&51,3#)0/& &

5014#//#,-/&2-3,14+"#,-&

$1,6(/& 7"+-8+18&'NO&./01&

>,)0/& 7"+-8+18&'NO&'52&'**0//&

&

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The next list of Application users is created during the

setup of the contact center server and therefore will not

need to be create:

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!""#$%&'$()*+,-.*

+,-.*/0* /)1(.2&'$()*

!

"#$%&'('!!

)%%*&+$#&,-!./01!2-3,14$#&,-!

510/0-+0!61,7%! 8#$-9$19!510/0-+0!:1,7%!

)++0%#!510/0-+0!87;/+1&%#&,-! <!

)++0%#!=7#>,3>9&$*,:!?@A@?! <!

)++0%#!.-/,*&+&#09!<,#&3&+$#&,-! <!

)++0%#!?0%*$+0/!B0$901! <!

C0D&+0!2-3,14$#&,-!

E,-#1,**09!C0D&+0/! ))('F'G!))('F'H!))('F'I!))('F'J!))('F'K!))!2EC!

E)5A!2-3,14$#&,-!

)//,+&$#09!E)5A!51,3&*0/! !

5014&//&,-/!2-3,14$#&,-!

61,7%/! 8#$-9$19!EL2!@-$;*09!

?,*0/! 8#$-9$19!EL2!@-$;*09!

!!

14"#$%&'!!

)%%*&+$#&,-!./01!2-3,14$#&,-!

510/0-+0!61,7%! 8#$-9$19!510/0-+0!:1,7%!

)++0%#!510/0-+0!87;/+1&%#&,-! <!

)++0%#!=7#>,3>9&$*,:!?@A@?! <!

)++0%#!.-/,*&+&#09!<,#&3&+$#&,-! <!

)++0%#!?0%*$+0/!B0$901! <!

C0D&+0!2-3,14$#&,-!

E,-#1,**09!C0D&+0/! 8@5FF'HEIMCNCEE!8@5FFNFKOEMIJHO!8@5FFNFKOEMC)NJ!8@5FFNFKOENCEAK!

E)5A!2-3,14$#&,-!

)//,+&$#09!E)5A!51,3&*0/! !

5014&//&,-/!2-3,14$#&,-!

61,7%/! 8#$-9$19!EL2!@-$;*09!

?,*0/! 8#$-9$19!EL2!@-$;*09!

!!

!

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The following End Users will also need to be created with

the correct settings from the table:

!"#$%&'($)*+"#'"&'#,$

%&'($-.$ /'0'12+"'$

3456'($

*+"7(+00'#$

.'89:'&$

*+"7(+00'#$

;(+<90'&$

=&&+:9>7'#$

!?7'"&9+"&$

@(+41&$ A+0'&$

!"#$ %&&#$ '()&&*&+,-*.-/+$ $$ %&&#$01$)23!"#341567189$%&&#$3$4)--$

'581:87:$--;$(1:$<=67=$'581:87:$-24$(18>96:$

'581:87:$--;$(1:$<=67=$'581:87:$--;<'("$?:@010=57850A1$'581:87:$-24$(18>96:$

!"%$ %&&%$ '()&&#B-C%.*.--$ $$ %&&%$01$)23!"#341567189$

'581:87:$--;$(1:$<=67=$'581:87:$-24$(18>96:$

'581:87:$--;$(1:$<=67=$'581:87:$--;<'("$?:@010=57850A1$'581:87:$-24$(18>96:$

DE#$ #&&#$ '()&&*&+,-%CFB,$<)-DE#$

$$ #&&#$01$)23DE341567189$#&&#$3$4)--$

'581:87:$--;$(1:$<=67=$'581:87:$-24$(18>96:$

'581:87:$--;$(1:$<=67=$'581:87:$--;<'("$?:@010=57850A1$'581:87:$-24$(18>96:$

DE%$ #&&%$ '()&&*&+,-%.?*F$ $$ #&&%$01$)23DE341567189$

'581:87:$--;$(1:$<=67=$'581:87:$-24$(18>96:$

'581:87:$--;$(1:$<=67=$'581:87:$--;<'("$?:@010=57850A1$'581:87:$-24$(18>96:$

G7=8:@01$ $$ $$ $$ H$IA16$J$ $$ $$

KAL1$ #&&C$ $$ MAL1$ #&&C$01$)23DE341567189$

'581:87:$--;$(1:$<=67=$'581:87:$-24$(18>96:$

'581:87:$--;$(1:$<=67=$'581:87:$--;<'("$?:@010=57850A1$'581:87:$-24$(18>96:$

!

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Voicemail Integrations

1. In order to accomplish this we first need to perform

all the required setup in callmanager for voicemail.

We start first with creating the voicemail ports:

Time Saver - You can create the ports manually or through the wizard.

Obviously the wizard is designed to save time as it creates the line group

for you will all the voicemail ports as members.

Below is the line group configuration:

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Next you will need to add this line group to a hunt list:

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Now we need to create the voicemail pilot (Hunt Pilot)

and point it to the correct hunt list:

We can now create the MWI on and MWI off extensions:

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We can now set the default voicemail pilot and

profile:

!Tip - You will see that we create a profile named strip. This will be used later

in task 5 in order to strip the “*” out of the extension before it rolls to

voicemail. The settings of this profile are shown below.

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Now in order to start the integration with Unity

Connections we will first need to create a port group:

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Next will need to define the callmanager servers Unity

will use through the Edit->Servers page under the port

group:

During the port group creation we were asked how many

ports and the device name prefix. This created the

ports for us automatically:

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Now we can finish the integration by settings the

parameters for the phone system to include the AXL

servers for importing users from callmanager:

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! Tip - This is also where the loop detection is enabled which is required in the

next task.

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Now we can define the AXL servers through the Edit->

Cisco Unified Communications Manager AXl Servers under

the phone system:

!Note - You will see the application user “Unity” which we create previously.

This user must be in the “Standard AXL User Group” that we created in order

to import users from callmanager.

2. In Unity Connections loop prevention is on by default.

However, you should be aware that there are two ways to

accomplish loop prevention. One is by extension and the

other is through DTMF tone based recognition.

3. In order to accomplish this task you must first modify

the default voicemail user template to change the default

password. You can then import all the users into Unity:

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You can now go to the import users page an import all

users:

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When finished your users page should look like the

following:

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! Tip - If the primary extension of the user is not set then the user will not show

up for import.

4. In order to configure the auto attendant we will first

need to create the appropriate CTI route point in

callmanager and forward all calls to voicemail for this

CTI route point:

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Time Saver -You can see two CTI route points listed as the second one will

be used to record the greeting for the call handler in this task.

We can setup the Greetings Administrator access for HQ

phone 1 user through the call routing section:

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Once this is complete we can now setup the call

handler for the auto attendant:

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Next we can setup the caller input options as required

in the task:

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We now need to enable and setup the closed extension

to take a message:

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We also need to modify the standard greeting so that a

message is not taken:

We need to also assign the call handler owner so that

the HQ phone 1 user can access the call handler and

change the greeting via the Greetings Administrator:

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5. Part of this task has already been configured during the

phone setup section and callmanager voicemail section.

What we need to double check is that the voicemail

profile named “STRIP” is assigned to the BR1 phone 2

extension of “*2002”:

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6. In order to accomplish this task we first need to define

access to the CUE module, the sip-ua for MWI, and then

dial-peers for the correct number(s) that need to be

forwarded to CUE:

interface Service-Engine1/0

ip unnumbered FastEthernet0/0.230

service-module ip address 10.1.230.2 255.255.255.0

service-module ip default-gateway 10.1.230.1

ip route 10.1.230.2 255.255.255.255 Service-Engine1/0

sip-ua

mwi-server ipv4:10.1.230.2 expires 3600 port 5060 transport udp

dial-peer voice 3180 voip

destination-pattern 3180

b2bua

session protocol sipv2

session target ipv4:10.1.230.2

dtmf-relay rtp-nte

codec g711ulaw

no vad

!

dial-peer voice 3000 voip

destination-pattern 3000

b2bua

session protocol sipv2

session target ipv4:10.1.230.2

dtmf-relay rtp-nte

codec g711ulaw

no vad

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Time Saver - In order to save time we can go ahed and create the dial-peer

need for task 8.

You can now start the CUE initialization process by

session into the CUE module from the BR2 gateway with

the following command:

BR2#service-module service-engine 1/0 session

Once you have completed the setup the CUE module will

reboot and is ready for configuration via the web

interface or via command line. You will now need to

set the DTMF relay for the CUE module to match the CME

SIP setup:

ccn subsystem sip

gateway address "10.1.230.1"

dtmf-relay rtp-nte

mwi sip outcall sub-notify

end subsystem

You can test access by pressing the voicemail button

on any BR2 phone.

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7. You can accomplish this via the web interface or via the

command line. We will show through the command line:

username br2phone3 create username BR2Phone2 create username BR2Phone1 create username admin create

username BR2Phone2 phonenumber "3002"

username BR2Phone1 phonenumber "3001"

username BR2Phone2 phonenumberE164 "442076303002"

username BR2Phone1 phonenumberE164 "442076303001"

voicemail mailbox owner "BR2Phone1" size 21176 description "BR2Phone1 mailbox" end mailbox voicemail mailbox owner "BR2Phone2" size 21176 description "BR2Phone2 mailbox" end mailbox

! Tip - You can test access and MWI by leaving a message and making sure the

MWI is working for each mailbox.

8. Part of this task was completed in task 6 and now we just

need to define the auto attendant in CUE:

ccn trigger sip phonenumber 3000

application "autoattendant"

enabled

maxsessions 8

end trigger

9. The was already accomplished earlier during the call

routing section. You will just need to test!

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IPCC Express Voice Application

1. In order to accomplish this task you will need to

complete the normal IPCC Express setup. Once this is

done you will need to create a Call Control Group

based on the task requirements:

!Tip - If required you can click the “Show More...” button and set the partition,

calling search space, AAR group, and other settings that may be required. In

this case we would probably want to set things like the device pool and

location.

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2. To accomplish this task you will need to create a new

application and trigger using the default auto

attendant script:

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3. First we need to create the application and trigger:

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We now can modify the resources and assign the skills

required for task 5:

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! Tip - If you don’t see any resources in IPCC Express then you probably

haven’t assigned the users an IPCC extension in callmanager.

We now need to create the contact services queue with

the same name we referenced under the application:

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We now need to create the IP Agent Phone Service:

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We can then assign the IP phone service to the proper

phones:

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You should now be able to login to the IP Agent Phone

service and test calls into the queue.

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4. In order to accomplish this we can modify the default

script to place the call on hold and then off hold

during the queuing process:

5. This was accomplished in task 3 by assigning the users

skills and then modifying the queue to route based on

“Most Skilled”.

6. This is shown in the script example provided under

task 4.

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! Tip - You should get to know the “Get Reporting Statistics” function in IPCC

Express. The function will allow you to create checks for just about any task.

CallManager Voice Application

1. Part of this task was already completed when we created

the user in callmanager. We now need to create the IP

phone service:

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Next we can create the device profile:

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We need to make sure we subscribe the device profile

to the extension mobility phone service:

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We then need to enable the required phones for extension

mobility and subscribe them to the IP phone service for

extension mobility:

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Presence

1. The first step is to run through the setup for

Presence after the initial install. Once this is

complete we can setup the required SIP trunk in

callmanager and set the SIP publish trunk in the

callmanager service parameters:

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Now set the CUP Publish Trunk in the callmanager

service parameters:

We can now create the CUPC in callmanager:

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Make sure on all the phones and CUPC in callmanager

that the subscribe calling search space is set

accordingly:

Next we need to assign the capabilities in

callmanager:

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Once this is complete we now need to finish the

configuration of the presence server. We can start by

validating the setup via System->CUCM Publisher:

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You can now configure the incoming and outgoing ACL’s:

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We now need to configure the CM SIP Publish trunk

under the Presence->Settings:

We now need to configure the callmanager as a presence

gateway:

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We now need to modify the proxy services settings:

We can then start configuring the CUPC settings:

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Next we need to configure the Unity server as a

voicemail server and then configure the voicemail

profile:

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We can now set the proper profiles for the HQ phone 1

user:

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Next we need to configure the “Deskphone Control

Settings” so that the CUPC client can control the

phone:

! Tip - The phones that need to be controlled by the “CtiGw” user were defined

earlier during the Application User section.

! Tip - You will also want to make sure that the phone being controlled has an

owner assigned and is associated to the end user.

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We can then enable the HQ phone 1 user for desktop

phone control:

Next you will need to the proxy domain and restart the

presence and proxy services:

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!Tip - You will need to create a host file entry on the machine running CUPC

based on the domain name you select in the presence server parameters.

Example: “presence.ccbootcamp.com” pointing to 10.1.200.23.

2. We need to first enable Unified Client access within

Unity Connections:

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Next we need to setup the Unity account user name and

password in the CUPC client and check that all

services are up and running:

!Note - You can switch between modes and check services for each setup. Keep

in mind that the service for the mode your not in will show as down or not

configured.

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Chapter 2 - Lab 2

Brief Lab Overview:

This lab is designed to test your SIP CME knowledge. The

focus is to configure the different features that are

available on the SIP side of CME.

Basic Campus Design:

1. Configure VLANS and IP networks in all three sites

according to the following two tables:

!

VLAN HQ BR1 BR2

Servers 1 N/A N/A

Data 20 120 220

Voice 30 130 230

!

Network HQ BR1 BR2

Servers 10.1.200.0/24 N/A N/A

Data 10.YY.20.0/24 10.YY.120.0/24 10.YY.220.0/24

Voice 10.YY.30.0/24 10.YY.130.0/24 10.YY.230.0/24

2. Make sure you set all voice traffic to use the voice

VLAN for all sites.

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3. Configure phones on HQ site to have IP addresses

120-130. Use IOS DHCP on the HQ router to accomplish

this task.

4. Configure BR1 phones to have IP addresses of 130–140

using the IOS DHCP on the BR1 Router.

5. Configure BR2 phones to have IP addresses 210–220

using the IOS DHCP on the BR2 router.

6. Configure HQ router as an NTP master clock with

stratum 2 for local time zone in Los Angeles and

configure BR1 and BR2 to synchronize their clocks to

HQ. BR1 is in New York time zone and BR2 is in India.

CallManager and CallManager Express:

1. Register all HQ and BR1 phones according to the dial

plan table using SCCP loads. BR2 phones should use SIP

loads and register accordingly. Make sure the phones

will display calling names in every scenario presented

(internal and PSTN):

2. Configure directory number 3003 as a second line on

both phones in BR2. When a call comes into 3003 it

should ring on both phones and be answered by the

first user to respond.

3. Make sure you that when the initiator of a conference

call within the BR2 site drops that the conference

call drops.

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4. Make sure all phones display the same general

information aside from the line specific

configuration; this refers to HQ and BR1.

5. Allow BR2 phone 1 to pickup calls, ringing and on

hold, on BR2 phone 2 and vice versa. This should work

for internal and external calls.

1

6. Make sure that BR2 phone 1 only allows one call

before a busy condition is set. BR2 phone 2 should

allow 3 calls before a busy condition is set.

7. Allow call list presence indication on BR2 phone 1

and 2.

8. BR2 phone 1 and phone 2 should not be able to park a

connected call.

9. Both BR2 phone 1 and BR2 phone 2 are members of VIP

sales group and are required to take calls dialed to

442076303111. The calls are expected to ring the

phones in longest idle order. Both phones should ring

for 5 seconds and if the call is not answered it

should be forwarded to DN 3000.

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Voice Gateways and Signaling:

1. Configure the HQ router as a IOS MGCP gateway using T1

PRI and NI as the line parameters.

2. Configure BR1 router as a IOS H.323 gateway using T1

PRI and NI as the line parameters.

3. Configure BR2 as E1 PRI and make sure all BR2 numbers

are reachable.

4. Configure HQ router as a gatekeeper with the following

details:

Local zone name: voiceie

Domain name: ccbootcamp.com

Register CallManager and CME to the gatekeeper.

Register the CallManager with a tech-prefix of “1#”

and a zone called “CM”. Register CME with a tech-

prefix of “1#” and to a zone called “CME”.

Call Routing:

1. In every site, phone 1 should be allowed to dial

international and below numbers, phone 2 should only

be allowed to call long distance and below.

2. Configure the following dialing options for users in

HQ:

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7+4 digits Calls to BR2

911 and 9911 Emergency

9+7 digits

The first is in the range of

2 to 9

Local

9+1+10 digits

The first and the forth are

in the range of 2 to 9.

Long Distance

9+011+any number of digits

Please allow users to dial

the hash sign at the end of

the number in order to have

faster response

International

• Local calls from HQ should use local HQ gateway

and BR1 gateway as backup.

• Long distance calls from HQ should use local HQ

gateway as first choice and then BR1 gateway.

• Calls to BR1 PSTN numbers should be routed out

BR1 gateway (Toll Bypass), with fallback to local

HQ gateway.

• International calls to BR2 numbers, using the 7+4

digits pattern, should be routed via the

gatekeeper as VoIP and use the local HQ gateway

as backup.

• International calls to BR2 local numbers should

be routed via the gatekeeper and then via the

local HQ gateway.

• All other International calls should route out

the HQ gateway.

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3. Configure the following dialing options for users in

BR1:

7+4 digits Calls to BR2

911 and 9911 Emergency

9+7 digits

The first is in the range of 2

to 9

Local

9+1+10 digits

The first and the forth are in

the range of 2 to 9.

Long Distance

9+011+any number of digits

Please allow users to dial the

hash sign at the end of the

number in order to have faster

response

International

• Local calls from BR1 should use the local BR1 IOS

gateway and then the HQ gateway as a backup.

• Long distance calls from BR1 should use the BR1

gateway as a first choice and then the HQ gateway

as a backup.

• Calls to HQ PSTN numbers (Area Code+Prefix)

should be routed out the HQ gateway (Toll Bypass)

with fallback to the local BR1 gateway.

• International calls to BR2 numbers, using the 7+4

digits pattern, should be routed via the

gatekeeper as VoIP and use the BR1 gateway as

backup.

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• International calls to BR2 local numbers should

be routed via the gatekeeper and then via the

local HQ gateway.

• All other International calls should route out

the BR1 gateway.

4. Configure the following dialing options for users in

BR2:

7+4 digits Calls to HQ and BR1

999 Emergency

9+7 digits Local

9+0+8 digits Long Distance

9+00+any number of digits

Please allow users to dial

the hash sign at the end of

the number in order to have

faster response

International

• International calls to HQ and BR1 should use the

gatekeeper first and the local PRI as a backup.

Codec and Call Admission Control:

1. All calls within a site should use the G.711 codec.

2. All calls between sites should use the G.729 codec.

3. Allow two concurrent calls to and from BR1.

4. Allow three concurrent calls from HQ and BR1 to BR2.

High Availability Features:

1. Configure BR1 as an SRST router.

2. Make sure you only allow BR1 phone 2 to register via

SRST during a failure.

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3. Preserve all PSTN dialing with using 9 as the first

digit in the string.

4. When in SRST mode, the BR1 phone 2 is not allowed to

dial international calls.

5. Allow HQ and BR1 phones to call each other in times

when more then 2 concurrent calls are in process

between the sites.

Media Resources Management:

1. Configure conferencing for the HQ site to use hardware

based resources in the HQ gateway first then hardware

resources in the BR1 gateway.

2. Configure conferencing for the BR1 site to use

hardware based resources in the BR1 gateway first then

hardware resources in the HQ gateway.

3. Configure transcoding for the HQ site to use hardware

based resources in the HQ gateway first then hardware

resources in the BR1 gateway.

4. Configure transcoding for the BR1 site to use hardware

based resources in the BR1 gateway first then hardware

resources in the HQ gateway.

5. Configure MOH for the HQ site based on unicast and

streamed from the publisher CallManager.

6. Configure multicast MOH for BR1, based on G729 codec

from the subscriber, and streamed from the BR1

gateway.

7. Allow 5 simultaneous meet me conferences for the HQ

users, in the number range 1201-1205. Every meet me

conference should allow a maximum of ten participants.

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8. Make sure that when the initiator of an Ad-Hoc

conference hangs up, the conference is allowed to

continue as long as an internal user is still

connected.

QoS Features:

4. Configure the frame relay link between HQ and BR1 as

if it is 384k in bandwidth, make sure you enable MLP

LFI on it.

• VoIP signaling should have 5% of the line as

guarantied bandwidth.

• VoIP barrier should have 33% of the line as priority

bandwidth.

• All other traffic should be weighted fair queued.

5. Configure the frame relay link between HQ and BR2 as

if it is 1200k in bandwidth.

• VoIP signaling should have 60k of the line reserved

bandwidth.

• VoIP barrier should have 360k of the line priority

bandwidth.

• All other traffic should be weighted fair queued.

6. The BR1 gateway should send the signaling and RTP

traffic marked as CS3 and EF.

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Voicemail Integrations:

1. Configure voicemail for HQ and BR1 based on the Unity

Connections server using the following parameters:

• Pilot 1170, port 1171-1174, MWI on 1180, MWI off

1181.

2. Make sure you prevent Unity Connections from calling

to itself and creating call loops which consume the

available ports.

3. Configure a voicemail box for phones 1 and 2 in HQ and

BR1 with a phone password of 147369 for all users.

4. Configure an auto attendant for HQ with DN 1000. It

should have a customized greeting that allows dialing

an extension and walking through the users directory

by pressing 1. When users press 0 they should be

forwarded to extension 1001. During closed times,

assume normal weekday schedule, it should take a

message and store it in the HQ phone 1 mailbox.

5. Create a second line on BR1 phone 2 with extension

*2002. A call coming to the primary line should roll

to this line if the primary is busy. The user should

be able to use this line for voicemail access. Do not

use any alternate extensions.

6. Configure BR2 CME to use the local Unity Express as a

voicemail, using the following parameters:

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• Pilot 3180.

7. Create voicemail for BR2 phone 1 and 2. Use a blank

phone password for all users.

8. Create an auto attendant for BR2 that will work when

DN 3000 is dialed.

9. Make sure that calls from HQ can route via gatekeeper

and hit the auto attendant in BR2.

IPCC Express Voice Application:

1. Configure the IPCC on the Contact Center server to

work with the CallManager using the following

parameters (all passwords are “cisco” and all pin’s

are “12345”):

Jtapi username: jtapi

Resource manager user: rmjtapi

CTI ports: 1015 – 1019.

2. Configure the auto attendant script to answer when a

call arrives at 1020.

3. Configure an ICD queue that will be triggered when

dialing extension 1021. HQ phone 1 should be used as

an IP phone agent for user “HQ1” with the password

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“cisco”. BR1 phone 1 should be used as a IP phone

agent with user “BR1” with a password “cisco”.

4. Callers should hear music on hold while delayed in

queue.

5. Calls should be routed via the queue based on a skill

named “Sales”. HQ phone 1 should be skilled higher

than BR1 phone 1.

6. Before a call is routed to the queue a check should be

performed to see if any agents are logged into the

queue. If no agents are logged in then the call should

be routed to HQ phone 2.

CallManager Voice Application:

1. Create extension mobility for user “john” with a

password of “cisco”. Which will allow him to use

extension 1007 on HQ and BR1 phone 2 in each site.

Make sure that the extension mobility service is only

available on phone 2 in HQ and BR1.

Presence:

1. Integrate the Presence server with CallManager. Make

sure the HQ Phone 1 user can login via CUPC and

control the desktop phone. HQ Phone 1 user should also

be allowed to switch to softphone mode.

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2. Make sure HQ Phone 1 can see the voicemail in their

voicemail box through the CUPC client.

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Solutions:

Basic Campus Design

1. Start with “show vlan” and make sure you have all

VLANS defined in all devices. If not create them and

then add the IP interfaces to each device with the

correct IP subnet from the table provided:

Time Saver - In order to save time you should go ahead and configure the

layer 2 QoS in the HQ switch.

HQ:

HQ-3750#

HQ-3750#config t

Enter configuration commands, one per line. End with CNTL/Z.

HQ-3750(config)#vlan 20!! !

HQ-3750(config-vlan)#name Data

HQ-3750(config-vlan)#exit

HQ-3750(config)#vlan 30!! !

HQ-3750(config-vlan)#name Voice

HQ-3750(config-vlan)#exit

HQ-3750(config)#

HQ-3750(config)#interface range FastEthernet 1/0/6 - 7

HQ-3750(config-if-range)#switchport voice vlan 30

HQ-3750(config-if-range)#switchport access vlan 20

HQ-3750(config-if-range)#switchport trunk encapsulation dot1q

HQ-3750(config-if-range)#switchport mode trunk

HQ-3750(config-if-range)#auto qos voip cisco-phone

HQ-3750(config)#interface range FastEthernet 1/0/1 - 5

HQ-3750(config-if-range)#switchport host

HQ-3750(config-if-range)#auto qos voip trust

HQ-3750(config)#interface FastEthernet 1/0/24

HQ-3750(config-if)#auto qos voip trust

HQ-3750(config)#interface FastEthernet0/0.20

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HQ-3750(config-if)#encapsulation dot1Q 20

HQ-3750(config-if)#ip address 10.1.20.1 255.255.255.0

HQ-3750(config)#exit

HQ-3750(config)#interface FastEthernet0/0.30

HQ-3750(config-if)#encapsulation dot1Q 30

HQ-3750(config-if)#ip address 10.1.30.1 255.255.255.0

! Tip - Don’t forget to add the new networks into the routing protocol.

HQ-3750(config)#router ospf 1

HQ-3750(config-router)#network 10.1.20.0 0.0.0.255 area 0

HQ-3750(config-router)#network 10.1.30.0 0.0.0.255 area 0

BR1:

BR1#vlan database

BR1(vlan)#vlan 120 name Data state active

VLAN 120 added:

Name: Data

State ACTIVE

BR1(vlan)#vlan 130 name Voice state active

VLAN 130 added:

Name: Voice

State ACTIVE

BR1(vlan)#exit

APPLY completed.

Exiting....

! Tip - The “vlan database” command has been deprecated so you can now use

the “vlan” command under global configuration mode as well.

BR1#config t

Enter configuration commands, one per line. End with CNTL/Z.

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BR1(config)#int vlan 120

BR1(config-if)#ip address 10.1.120.1 255.255.255.0

BR1(config-if)#description Data

BR1(config-if)#int vlan 130

BR1(config-if)#ip address 10.1.130.1 255.255.255.0

BR1(config-if)#description Voice

BR1(config-if)#exit

BR1(config)#interface range FastEthernet 0/3/0 - 1

BR1(config-if-range)#switchport voice vlan 130

BR1(config-if-range)#switchport access vlan 120

BR1(config-if-range)#no shut

! Tip - Don’t forget to add the new networks into the routing protocol.

BR1(config)#router ospf 1

BR1(config-router)#network 10.1.120.0 0.0.0.255 area 0

BR1(config-router)#network 10.1.130.0 0.0.0.255 area 0

BR2:

BR2#vlan database

BR2(vlan)#vlan 220 name Data state active

VLAN 220 added:

Name: Data

State ACTIVE

BR2(vlan)#vlan 230 name Voice state active

VLAN 230 added:

Name: Voice

State ACTIVE

BR2(vlan)#exit

APPLY completed.

Exiting....

BR2#config t

Enter configuration commands, one per line. End with CNTL/Z.

BR2(config)#int vlan 220

BR2(config-if)#ip address 10.1.220.1 255.255.255.0

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BR2(config-if)#description Data

BR2(config-if)#int vlan 230

BR2(config-if)#ip address 10.1.230.1 255.255.255.0

BR2(config-if)#description Voice

BR2(config-if)#exit

BR2(config)#interface range FastEthernet 0/1/0 - 1

BR2(config-if-range)#switchport voice vlan 230

BR2(config-if-range)#switchport access vlan 220

BR2(config-if-range)#no shut

! Tip - Don’t forget to add the new networks into the routing protocol.

BR2(config)#router ospf 1

BR2(config-router)#network 10.1.220.0 0.0.0.255 area 0

BR2(config-router)#network 10.1.230.0 0.0.0.255 area 0

2. Validate your configurations with the “show

vlan” ( or “show current” under the vlan database)

and “show cdp neighbors” commands.

!

HQ-3750#show vlan

VLAN Name Status Ports

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

1 default active ! Fa1/0/1, Fa1/0/2, Fa1/0/3, Fa1/0/4,

! ! ! ! ! ! Fa1/0/5, Fa1/0/8, Fa1/0/9, Fa1/0/10,

! ! ! ! ! ! Fa1/0/11, Fa1/0/12, Fa1/0/13, Fa1/0/14,

! ! ! ! ! ! Fa1/0/15, Fa1/0/16, Fa1/0/17, Fa1/0/18,

! ! ! ! ! ! Fa1/0/19, Fa1/0/20, Fa1/0/21, Fa1/0/22,

! ! ! ! ! ! Fa1/0/23, Gi1/0/1, Gi1/0/2

20 Data active

30 Voice active

1002 fddi-default act/unsup

1003 token-ring-default act/unsup

1004 fddinet-default act/unsup

1005 trnet-default act/unsup

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HQ-3750#show cdp neigh

Capability Codes: R - Router, T - Trans Bridge, B - Source Route Bridge

S - Switch, H - Host, I - IGMP, r - Repeater, P - Phone

Device ID Local Intrfce Holdtme Capability Platform Port ID

CCMSUB Fas 1/0/1 152 H none foundeth0

CCMPUB Fas 1/0/1 157 H none foundeth0

HQ_Rtr Fas 1/0/24 140 R S I 2811 Fas 0/0.1

uccx. Fas 1/0/5 172 H Win2000 SeEth 1/1

presence Fas 1/0/4 145 H none foundeth0

SEP003094C27864 Fas 1/0/6 138 H P IP Phone 7Port 1

SEP003094C2DA38 Fas 1/0/7 132 H P IP Phone 7Port 1

SEP0002FDAEFDDF Fas 1/0/8 159 H P IP Phone 7Port 1

Unity Fas 1/0/3 162 H none foundeth0

PSTN-WAN Fas 1/0/22 124 R S I 2811 Fas 0/0

HQ-3750#

!Note - The publisher and subscriber are on the same physical box so they will

show as if they are connected to the same port on the switch. This is normal

and you should configure FastEthernet 1/0/2 as if the subscriber is plugged

into that port.

BR1#vlan database

BR1(vlan)#show current

VLAN ISL Id: 1

Name: default

Media Type: Ethernet

VLAN 802.10 Id: 100001

State: Operational

MTU: 1500

Translational Bridged VLAN: 1002

Translational Bridged VLAN: 1003

VLAN ISL Id: 120

Name: Data

Media Type: Ethernet

VLAN 802.10 Id: 100220

State: Operational

MTU: 1500

VLAN ISL Id: 130

Name: Voice

Media Type: Ethernet

VLAN 802.10 Id: 100230

State: Operational

MTU: 1500

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BR1#show cdp neigh

Capability Codes: R - Router, T - Trans Bridge, B - Source Route Bridge

S - Switch, H - Host, I - IGMP, r - Repeater

Device ID Local Intrfce Holdtme Capability Platform Port ID

HQ_Rtr Ser 0/1/0.1 161 R S I 2811 Ser 0/2/0.1

SEP0016C72D3DCC Fas 0/3/1 146 H IP Phone Port 1

SEP003094C3DCF9 Fas 0/3/0 143 H IP Phone Port 1

BR1#

BR2# show cdp neigh

Capability Codes: R - Router, T - Trans Bridge, B - Source Route Bridge

S - Switch, H - Host, I - IGMP, r - Repeater

Device ID Local Intrfce Holdtme Capability Platform Port ID

HQ_Rtr Ser 0/2/0.1 133 R S I 2811 Ser 0/2/0.2

SEP001F6C7FD6C1 Fas 0/1/1 167 H IP Phone Port 1

SEP001F6C7F722B Fas 0/1/0 167 H IP Phone Port 1

BR2#

BR2#vlan database

BR2(vlan)#show current

VLAN ISL Id: 1

Name: default

Media Type: Ethernet

VLAN 802.10 Id: 100001

State: Operational

MTU: 1500

Translational Bridged VLAN: 1002

Translational Bridged VLAN: 1003

VLAN ISL Id: 220

Name: Data

Media Type: Ethernet

VLAN 802.10 Id: 100220

State: Operational

MTU: 1500

VLAN ISL Id: 230

Name: Voice

Media Type: Ethernet

VLAN 802.10 Id: 100230

State: Operational

MTU: 1500

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3. In order to accomplish this we need to first create

the dhcp excluded address and then define the DHCP

pool:

HQ_Rtr(config)#ip dhcp excluded-address 10.1.30.1 10.1.30.119

HQ_Rtr(config)#ip dhcp excluded-address 10.1.30.131 10.1.30.255

HQ_Rtr(config)#ip dhcp pool IPPHONE

HQ_Rtr(config-config)#network 10.1.30.0 255.255.255.0

HQ_Rtr(config-config)#default-router 10.1.30.1

HQ_Rtr(config-config)#option 150 ip 10.1.200.21 10.1.200.25

Time Saver - In order to save time you should use notepad to copy and

paste configurations which may be the same or need little modification.

4. We can use the same configuration from the last task

but modify for BR1:

BR1(config)#ip dhcp excluded-address 10.1.130.1 10.1.130.129

BR1(config)#ip dhcp excluded-address 10.1.130.141 10.1.130.255

BR1(config)#ip dhcp pool IPPHONE

BR1(config-config)#network 10.1.130.0 255.255.255.0

BR1(config-config)#default-router 10.1.130.1

BR1(config-config)#option 150 ip 10.1.200.21 10.1.200.25

5. Perform the same for BR2:

BR2(config)#ip dhcp excluded-address 10.1.230.1 10.1.230.209

BR2(config)#ip dhcp excluded-address 10.1.230.221 10.1.230.255

BR2(config)#ip dhcp pool IPPHONE

BR2(config-config)#network 10.1.230.0 255.255.255.0

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BR2(config-config)#default-router 10.1.230.1

BR2(config-config)#option 150 ip 10.1.230.1

6. Next configure each router for the appropriate time

zone and NTP settings:

HQ_Rtr(config)#clock timezone PST -8

HQ_Rtr(config)#clock summer-time PDT recurring

HQ_Rtr(config)#ntp master 2

BR1(config)#clock timezone EST -5

BR1(config)#clock summer-time EDT recurring

BR1(config)#ntp server 10.10.32.1

BR2(config)#clock timezone UTC +5 30

BR2(config)#clock summer-time PDT recurring

BR2(config)#ntp server 10.10.32.1

! Tip - Use can use your candidate PC to look up the time zone of most

countries but most likely they will specify for you.

! Tip - Don’t forget to set the time zone settings in CallManager as it may not be

mentioned in the phone setup tasks.

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CallManager and CallManager Express:

1. This tasks requires a lot of configuration as this is

the base line for everything you do in CallManager. We

will setup everything from device pools, locations,

regions, media resource lists and groups, route lists

and route groups, partitions and calling search

spaces.

!

Lets start off with service activation first as

nothing will work if we don’t have services activated:

Make sure you enable services on both servers and all

services except the “Cisco Messaging Interface”.

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!Tip - Enabling all services as stated above will ensure that all needed services

are activated. Unless you are specifically told to only enable needed services

then this strategy will allow you to make sure all the required services are

running.

Next we need to start configuring the required

CallManager settings needed to allow for the

registration of HQ and BR1 phones.

Make sure the proper servers are defined:

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Next create a CallManager group that will allow for

devices to register to the subscriber and then the

publisher:

When finished you should see the following to groups:

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! Note - The requirement for registration order is normally on the first page of

instructions.

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Next we need to create the regions:

We need to make sure the settings between the regions

are set appropriately:

Time Saver - This is where reading through the whole lab will save a lot of

time as we can set the proper region settings and never return to this page.

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Now we need to create the HQ and BR1 Date/Time groups:

! Note - This is also where we could set the date/time format if asked during a

specific task.

Next we need to create the locations:

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Time Saver - This is where reading through the whole lab will save a lot of

time as we can set the proper location settings and never return to this page.

We now need to create the media resource groups and

lists:

!Tip - In reading the lab we know that there is a need to separate the MOH

servers and it is always a good idea to separate hardware from software

resources.

We can then create the needed media resource lists:

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Below is the full list of the media resource lists:

We now need to create the device pools:

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Time Saver - We know from reading the lab that one phone will not be

allowed to register in SRST. We can create the device pool for this function

now in order to save time later.

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Here is the final list of device pools:

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Next we need to check the Enterprise Parameters and

make changes as needed:

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!Tip - Changing the host name to an IP address in the URL settings will prevent

authentication and IP phone service failures. This is highly recommended

whether you have a DNS server or not!

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We can also create the AAR groups now as this will

help to cut down the number of times we need to visit

the phone configuration page:

Once we have both groups created we need to set the

dial prefix according to the dial plan:

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Now we will need to create the the partitions and

calling search spaces that will be needed for the lab:

On the next page is a table of calling search spaces

and the partitions that belong to each calling search

space.

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!"##$%&'()"*+,'(-"+)'

."/)' 0)1+*$-2$3%' 4352)'6"*2$2$3%1'73*'2,$1'!"##$%&'()"*+,'(-"+)'

!""#$%&#''%( !""#$%&#''%( )*+$%&+''%(

!""#$%&#,-*( !""#$%&#,-*( )*+$%&+.&&(

)*+$%&+,/012/34(

)*+$%&+*566$7)'""(

)*+89+,/012/34(

)*+$%:+,/012/34(

)*+$%&+6;<34(

)*+$%&+6=(

)*+$%&+,-*(

!""#$%&#,-*>%-'6( !""#$%&#,-*>%-'6( )*+$%&+.&&(

)*+$%&+,/012/34(

)*+$%&+*566$7)'""(

)*+89+,/012/34(

)*+$%:+,/012/34(

!""#$%&#6=( !""#$%&#6=( )*+$%&+.&&(

)*+$%&+,/012/34(

)*+$%&+*566$7)'""(

)*+89+,/012/34(

)*+$%:+,/012/34(

)*+$%&+6;<34(

)*+$%&+6=(

!""#$%&#65!'6( !""#$%&#65!'6( )*+$%&+.&&(

)*+$%&+,/012/34(

)*+$%&+*566$7)'""(

)*+89+,/012/34(

)*+$%:+,/012/34(

)*+$%&+6;<34(

!""#$%:( !""#$%:( )*+$%&+,/012/34(

)*+$%:+,/012/34(

)*+89+,/012/34(

)*+$%:+*566$7)'""(

!""#89#''%( !""#89#''%( )*+89+''%(

!""#89#,-*( !""#89#,-*( )*+89+.&&(

)*+89+,/012/34(

)*+89+*566$7)'""(

)*+$%&+,/012/34(

)*+$%:+,/012/34(

)*+89+6;<34(

)*+89+6=(

)*+89+,-*(

!""#89#,-*>%-'6( !""#89#,-*>%-'6( )*+89+.&&(

)*+89+,/012/34(

)*+89+*566$7)'""(

)*+$%&+,/012/34(

)*+$%:+,/012/34(

!""#89#6=( !""#89#6=( )*+89+.&&(

)*+89+,/012/34(

)*+89+*566$7)'""(

)*+$%&+,/012/34(

)*+$%:+,/012/34(

)*+89+6;<34(

)*+89+6=(

!""#89#65!'6( !""#89#65!'6( )*+89+.&&(

)*+89+,/012/34(

)*+89+*566$7)'""(

)*+$%&+,/012/34(

)*+$%:+,/012/34(

)*+89+6;<34(

!""#?5,!>@',6( !""#?5,!>@',6( )*+$%&+,/012/34(

)*+89+,/012/34(

(

(

We should now be able to add the phones per the

requirements in the table.

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HQ Phone 1:

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HQ Phone 2:

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BR1 Phone 1:

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BR1 Phone 2:

!Note - In reading the lab we know the BR1 Phone 2 requires another extension

“*2002”. We can create that as part of the initial phone setup which we have

done here or you could wait till that specific task.

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BR1 Phone 2 (Line 2):

! Tip - Most of the phone settings per phone are the same per site except for the

calling search space.

Time Saver - In order to save time you can use auto registration on the

callmanager and then modify each phone to the correct settings.

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We can now start the registration of phones within the

BR2 site. We will first need to define all the

required settings underneath the “telephony-service”

and “voice register global” configuration. We can then

create the phones and their associated DN’s:

BR2(config)#telephony-service

BR2(config-telephony)#authentication credential admin cisco

BR2(config-telephony)#max-ephones 2

BR2(config-telephony)#max-dn 5

BR2(config-telephony)#ip source-address 10.1.230.1 port 2000

BR2(config-telephony)#voicemail 3180

BR2(config-telephony)#max-conferences 8 gain -6

BR2(config-telephony)#call-park system application

BR2(config-telephony)#call-forward pattern .T

BR2(config-telephony)#dn-webedit

BR2(config-telephony)#time-webedit

BR2(config-telephony)#transfer-system full-blind

BR2(config-telephony)#transfer-pattern 9.T

BR2(config-telephony)#fac standard

BR2(config-telephony)#create cnf-files

BR2(config-telephony)#exit

BR2(config)#voice register global

BR2(config-register-global)#mode cme

BR2(config-register-global)#source-address 10.1.230.1 port 5060

BR2(config-register-global)#max-dn 10

BR2(config-register-global)#max-pool 2

BR2(config-register-global)#load 7941 SIP41.8-3-3S

BR2(config-register-global)#mwi stutter

BR2(config-register-global)#mwi reg-e164

BR2(config-register-global)#voicemail 3180

BR2(config-register-global)#tftp-path flash:

BR2(config-register-global)#create profile

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Now we can configure the phones and their DN’s:

BR2(config)#voice register dn 1

BR2(config-register-dn)#number 3001

BR2(config-register-dn)#call-forward b2bua noan 3180 timeout 20

BR2(config-register-dn)#pickup-call any-group

BR2(config-register-dn)#pickup-group 1

BR2(config-register-dn)#name BR2 Phone 1

BR2(config-register-dn)#mwi

BR2(config-register-dn)#voice register dn 2

BR2(config-register-dn)#number 3002

BR2(config-register-dn)#call-forward b2bua noan 3180 timeout 20

BR2(config-register-dn)#pickup-call any-group

BR2(config-register-dn)#pickup-group 1

BR2(config-register-dn)#name BR2 Phone 2

BR2(config-register-dn)#mwi

We can also configure the extra DN specified for the

shared line on BR2 phone 1 and 2:

BR2(config-register-dn)#voice register dn 3

BR2(config-register-dn)#number 3003

BR2(config-register-dn)#shared-line max-calls 6

We can now configure the phones:

BR2(config)#voice register pool 1

BR2(config-register-pool)#id mac 001F.6C7F.D6C1

BR2(config-register-pool)#type 7941

BR2(config-register-pool)#number 1 dn 1

BR2(config-register-pool)#number 2 dn 3

BR2(config-register-pool)#dtmf-relay rtp-nte

BR2(config-register-pool)#codec g711ulaw

BR2(config-register-pool)#no keep-conference

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BR2(config-register-pool)#voice register pool 2

BR2(config-register-pool)#id mac 001F.6C7F.722B

BR2(config-register-pool)#type 7941

BR2(config-register-pool)#number 1 dn 2

BR2(config-register-pool)#number 2 dn 3

BR2(config-register-pool)#dtmf-relay rtp-nte

BR2(config-register-pool)#codec g711ulaw

BR2(config-register-pool)#no keep-conference

Next we need to allow the SIP firmware to be

downloaded from the router flash:

BR2(config)# tftp-server flash:SIP41.8-4-4S.loads

BR2(config)# tftp-server flash:term41.default.loads

BR2(config)# tftp-server flash:SIP41.8-3-3S.loads

BR2(config)# tftp-server flash:jar41sip.8-3-2-27.sbn

BR2(config)# tftp-server flash:dsp41.8-3-2-27.sbn

BR2(config)# tftp-server flash:cvm41sip.8-3-2-27.sbn

BR2(config)# tftp-server flash:cnu41.8-3-2-27.sbn

BR2(config)# tftp-server flash:apps41.8-3-2-27.sbn

!Tip - If you like to use the CME and CUE web interface the the following

commands should be entered at this stage to prepare you to use the web

interface.

BR2(config)#ip http server

BR2(config)#no ip http secure-server

BR2(config)#ip http path flash:

BR2(config)#telephony-service

BR2(config-telephony)#web admin system name admin password cisco

BR2(config-telephony)#url services http://10.1.230.2/voiceview/common/

login.do (Inserted by default in most cases)

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BR2(config-telephony)#url authentication http://10.1.230.1/CCMCIP/

authenticate.asp (Inserted by default in most cases)

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2. This was already completed in the above solution with the

“voice register dn 3” configuration and the addition of

this button to each “voice register pool”.

3. This was already completed in the above solution with the

“no keep-conference” command under the “voice register

pool” configuration.

4. This was accomplished in task 1 by setting the alerting

name, caller id, external phone number mask, and date/

time group format the same in CallManager.

5. This was accomplished in task 1 by setting setting the

“pickup-group 1” command under all the DN’s.

6. This can be accomplished through adding the following

command to the proper voice register pool:

BR2(config)#voice register pool 1

BR2(config-voice-register)#busy-trigger-per-button 2

BR2(config)#voice register pool 2 BR2(config-voice-register)#busy-trigger-per-button 3

7. The is accomplished by enabling presence on the BR2

gateway, setting the voice register DN’s to allow

watching, and then adding the presence call-list command

to the voice register pool:

BR2(config)#presence BR2(config-presence)#presence call-list

BR2(config)#voice register dn 1

BR2(config-voice-register)#allow watch

BR2(config)#voice register dn 2

BR2(config-voice-register)#allow watch

BR2(config)#voice register pool 1

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BR2(config-voice-register)#presence call-list

BR2(config)#voice register pool 2 BR2(config-voice-register)#presence call-list

8. This is accomplished through a voice register template

that is then assigned to each voice register pool:

BR2(config)#voice register template 1 BR2(config-register)#softkeys connected Confrn Endcall Hold Trnsfer

BR2(config)#voice register pool 1

BR2(config-voice-register)#template 1

BR2(config)#voice register pool 2 BR2(config-voice-register)#template 1

9. This is accomplished with the following configuration:

BR2(config)#voice hunt-group 1 sequential

BR2(config-voice-hunt-group)#final 3000

BR2(config-voice-hunt-group)#list 3001,3002

BR2(config-voice-hunt-group)#timeout 10

BR2(config-voice-hunt-group)#pilot 3111

!Note - Though the task asks us to allow the number to be reach by the full

E164 number we can perform this action later when setting up the call routing

as all calls will need to be striped to four digits.

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Voice Gateways and Signaling

1. In order to enable the HQ router as an MGCP gateway we

need the following configuration on the HQ router:

HQ_Rtr(config)#network-clock-participate wic 0

HQ_Rtr(config)#isdn switch-type primary-ni

HQ_Rtr(config)#voice-card 0

HQ_Rtr(config-voicecard)#dspfarm

HQ_Rtr(config-voicecard)#dsp services dspfarm

HQ_Rtr(config-voicecard)#exit

HQ_Rtr(config)#controller T1 0/0/0

HQ_Rtr(config-controller)#cablelength long 0db

HQ_Rtr(config-controller)#pri-group timeslots 1-3,24 service mgcp

HQ_Rtr(config-controller)#exit

HQ_Rtr(config)#interface Serial0/0/0:23

HQ_Rtr(config-if)# isdn bind-l3 ccm-manager

HQ_Rtr(config-if)#exit

HQ_Rtr(config)#ccm-manager redundant-host 10.1.200.21

HQ_Rtr(config)#ccm-manager mgcp

HQ_Rtr(config)#mgcp

HQ_Rtr(config)#mgcp call-agent 10.1.200.25 service-type mgcp version 0.1

HQ_Rtr(config)#mgcp fax t38 ecm

HQ_Rtr(config)#mgcp

! Note - In IOS release 12.4 the dial-peer no longer needs to be created with the

service command.

Time Saver - If you have multiple MGCP or H.323 gateways then use

notepad to copy and paste common configuration components.

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Next we will need to configure the gateway in

CallManager:

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!Tip - If you have set or been asked to set a domain name on the router the

device name for MGCP would be the router name plus the domain name. So in

our case it could be “HQ_Rtr.voiceie.com” if we had “voiceie.com” as a

domain name on the router.

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2. In order to enable the BR1 router as an H.323 gateway we

need the following configuration on the BR1 router:

BR1(config)#network-clock-participate wic 1

BR1(config)#isdn switch-type primary-ni

BR1(config)#voice-card 0

BR1(config-voicecard)#dsp services dspfarm

BR1(config-voicecard)#exit

BR1(config)#voice translation-rule 1

BR1(cfg-translation-rule)#rule 1 /\(^702333\)\(2...\)/ /\2/

BR1(cfg-translation-rule)#exit

BR1(config)#voice translation-profile STRIP

BR1(cfg-translation-profile)#translate called 1

BR1(cfg-translation-profile)#exit

BR1(config)#voice-port 0/1/0:23

BR1(config-voiceport)#translation-profile incoming STRIP

BR1(config-voiceport)#exit

BR1(config)#controller T1 0/1/0

BR1(config-controller)#framing esf

BR1(config-controller)# linecode b8zs

BR1(config-controller)# pri-group timeslots 1-3,24

BR1(config-controller)# exit

BR1(config)#interface Loopback0

BR1(config-if)#h323-gateway voip interface

BR1(config-if)#h323-gateway voip bind srcaddr 10.10.32.2

BR1(config-if)#exit

BR1(config)#interface Serial0/1/0:23

BR1(config-if)#isdn outgoing display-ie

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Next we will need to setup BR1 as an H.323 gateway in

CallManager:

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!Tip - The most common issue with H.323 gateways is that IP address specified

in CallManager is not the IP address that the H.323 process in the router is

binding. Make sure these IP address match!

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3. In order to enable the BR2 router as an H.323 gateway we

need the following configuration on the BR2 router:

BR2(config)#network-clock-participate wic 0

BR2(config)#isdn switch-type primary-net5

BR2(config)#voice translation-rule 1

BR2(cfg-translation-rule)#rule 1 /\(^44207630\)\(3...\)/ /\2/

BR2(cfg-translation-rule)#exit

BR2(config)#voice translation-profile STRIP

BR2(cfg-translation-profile)#translate called 1

BR2(cfg-translation-profile)#exit

BR2(config)#voice-card 0

BR2(config-voicecard)#dspfarm

BR2(config-voicecard)#dsp services dspfarm

BR2(config-voicecard)#exit

BR2(config)#controller E1 0/0/0

BR2(config-controller)#pri-group timeslots 1-3,16

BR2(config-controller)#exit

BR2(config)#interface Serial0/0/0:15

BR2(config-if)#isdn outgoing display-ie

BR2(config-if)#exit

BR2(config)#voice-port 0/0/0:15

BR2(config-voiceport)#translation-profile incoming STRIP

4. In order to setup the gatekeeper we need to perform the

following configuration on the HQ router:

HQ_Rtr(config)#gatekeeper

HQ_Rtr(config-gk)#zone local CM ccbootcamp.com 10.10.32.1

HQ_Rtr(config-gk)#zone local CME ccbootcamp.com

HQ_Rtr(config-gk)#gw-type-prefix 1#* default-technology

HQ_Rtr(config-gk)#no shutdown

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Next we need to enable CallManager to register to the

gatekeeper through a H.225 gatekeeper controlled

trunk. We can accomplish this by adding the gatekeeper

and then the trunk in CallManager:

!Tip - If you would like to lower the time it takes for gatekeeper to register

after a loss of connectivity then change the “Registration Retry Timeout” to 30

seconds.

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Now we need to add the trunk:

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! Tip - You can validate registration with the “show gatekeeper endpoints”

command on the HQ router.

HQ_Rtr#show gatekeeper endpoints

GATEKEEPER ENDPOINT REGISTRATION

================================

CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags

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

10.1.200.21 33072 10.1.200.21 32794 CM H323-GW

H323-ID: BR2_1

Voice Capacity Max.= Avail.= Current.= 0

10.1.200.25 33077 10.1.200.25 32785 CM H323-GW

H323-ID: BR2_2

Voice Capacity Max.= Avail.= Current.= 0

Total number of active registrations = 4

HQ_Rtr#

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Next we will need to configure the BR2 router is

register with the gatekeeper. This requires the

following configuration:

BR2(config)#interface Loopback0

BR2(config-if)#h323-gateway voip interface

BR2(config-if)#h323-gateway voip id CME ipaddr 10.10.32.1 1719

BR2(config-if)#h323-gateway voip h323-id CME

BR2(config-if)#h323-gateway voip tech-prefix 1#

BR2(config-if)#h323-gateway voip bind srcaddr 10.10.32.3

BR2(config-if)#exit

BR2(config)#gateway

! Tip - Once again you can verify with the “show gatekeeper endpoints”

command on the HQ router.

HQ_Rtr#show gatekeeper endpoints

GATEKEEPER ENDPOINT REGISTRATION

================================

CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags

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

10.10.32.3 1720 10.10.32.3 57205 CM H323-GW

H323-ID: CME

Voice Capacity Max.= Avail.= Current.= 0

10.1.200.21 33072 10.1.200.21 32794 CM H323-GW

H323-ID: BR2_1

Voice Capacity Max.= Avail.= Current.= 0

10.1.200.25 33077 10.1.200.25 32785 CM H323-GW

H323-ID: BR2_2

Voice Capacity Max.= Avail.= Current.= 0

Total number of active registrations = 4

HQ_Rtr#

!Note - You will notice that the directory numbers from the SIP CME did not

register with gatekeeper. This is to be expected as SIP CME phones cannot be

auto registered to gatekeeper.

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Call Routing

1. The has already been accomplished for HQ and BR1 when

we setup the phones previously by giving them the

proper calling search space. To enable class of

service for the BR2 site we need to create the

following COR list:

dial-peer cor custom

name Local

name LD

name INT

name Emergency

dial-peer cor list Emergency

member Emergency

dial-peer cor list Local

member Local

dial-peer cor list Int

member INT

dial-peer cor list LD

member LD

dial-peer cor list Phone1

member Local

member LD

member INT

member Emergency

dial-peer cor list Phone2

member Local

member LD

member Emergency

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Now that we have created the COR list we need to

assign the proper COR list to the voice register pool

of the proper phone:

BR2(config)#voice register pool 1

BR2(config-register-pool)#cor incoming Phone1 default

BR2(config-register-pool)#voice register pool 2

BR2(config-register-pool)#cor incoming Phone2 default

2. In order to accomplish the next two tasks we need to

create all the route groups and route lists first. We

will first need to create the route groups:

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Then next the route lists in the following tables with

the proper settings:

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!"#$%&'()$&

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!"#$%&'()$&

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A summary of all the route lists is below:

We can now create all the route patterns for the HQ

call routing and point them to the proper route list:

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! Note - All the digit manipulation and phone number masking is done in the

route lists and therefore should not be done in the route pattern.

Next we need to add the proper call routing to the

gatekeeper on the HQ gateway:

HQ_Rtr(config-gk)#zone prefix CME 3...

HQ_Rtr(config-gk)#zone prefix CM 7....

HQ_Rtr(config-gk)#zone prefix CME 96......

3. We can now use the previous created route groups and

route lists to setup the BR1 call routing by adding

the following route patterns:

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Now since the BR1 gateway is an H.323 gateway we need

to add the proper dial-peers to the BR1 gateway to

complete the call routing. We may also want to take

into account the SRST task for maintaining class of

service and create the COR list at the same time:

dial-peer cor custom

name Local

name LD

name INT

name Emergency

dial-peer cor list Emergency

member Emergency

dial-peer cor list Local

member Local

dial-peer cor list Int

member INT

dial-peer cor list LD

member LD

dial-peer cor list Phone1

member Local

member LD

member INT

member Emergency

dial-peer cor list Phone2

member Local

member LD

member Emergency

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We can now create the dial-peers with the COR already

configured:

dial-peer voice 1 pots

incoming called-number .

direct-inward-dial

dial-peer voice 2000 voip

preference 1

destination-pattern 2...

session target ipv4:10.1.200.25

ip qos dscp cs3 signaling

dial-peer voice 2001 voip

preference 2

destination-pattern 2...

session target ipv4:10.1.200.21

ip qos dscp cs3 signaling

dial-peer voice 7 pots

corlist outgoing Local

destination-pattern 9[2-9]......

port 0/1/0:23

forward-digits 7

dial-peer voice 11 pots

corlist outgoing LD

destination-pattern 91[2-9]..[2-9]......

port 0/1/0:23

forward-digits 11

!

dial-peer voice 911 pots

corlist outgoing Emergency

destination-pattern 911

port 0/1/0:23

forward-digits all

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!

dial-peer voice 9911 pots

corlist outgoing Emergency

destination-pattern 9911

port 0/1/0:23

forward-digits 3

!

dial-peer voice 9011 pots

corlist outgoing Int

destination-pattern 9011T

port 0/1/0:23

prefix 011

4. We can now setup the call routing for BR2 with the

following configuration on the BR2 router. We first

need to allow all the communications that maybe needed

in our environment:

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

supplementary-service h450.12

no supplementary-service sip refer

fax protocol cisco

h323

sip

registrar server expires max 1200 min 300

Next we need to create a voice class codec for our

calls through gatekeeper so that the call setup will

not fail:

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voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g729r8

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Now we can create all the needed dial-peers with the

appropriate COR list which were previously defined:

dial-peer voice 1 pots

incoming called-number .

direct-inward-dial

!

dial-peer voice 2 pots

corlist outgoing Local

destination-pattern 90[1-9].......

port 0/0/0:15

prefix 0

!

dial-peer voice 1000 voip

preference 1

destination-pattern 71...

voice-class codec 1

session target ras

!

dial-peer voice 1001 pots

preference 2

destination-pattern 71...

port 0/0/0:15

prefix 0017029461

!

dial-peer voice 2000 voip

preference 1

destination-pattern 72...

voice-class codec 1

session target ras

!

dial-peer voice 2001 pots

preference 2

destination-pattern 72...

port 0/0/0:15

prefix 0017023332

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!

dial-peer voice 7 pots

corlist outgoing Local

destination-pattern 9[1-9]......

port 0/0/0:15

forward-digits 7

!

dial-peer voice 999 pots

corlist outgoing Emergency

destination-pattern 999

port 0/0/0:15

forward-digits all

!

dial-peer voice 900 pots

corlist outgoing Int

destination-pattern 900.T

port 0/0/0:15

prefix 00

We also need to add a translation pattern to strip the

“7” as the call comes into callmanager from BR2:

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We will also need to enable transcoding on the BR2

gateway to support calls transgressing from HQ/BR1 to

BR2:

sccp local FastEthernet0/0.230

sccp ccm 10.1.230.1 identifier 1 priority 1 version 3.1

sccp

sccp ccm group 1

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associate ccm 1 priority 1

associate profile 1 register TRANSCODE

dspfarm profile 1 transcode

codec g711ulaw

codec g711alaw

codec g729r8

codec g729abr8

codec g729ar8

codec g729br8

maximum sessions 4

associate application SCCP

telephony-service

sdspfarm units 2

sdspfarm transcode sessions 4

sdspfarm tag 1 TRANSCODE

!Tip - At this point you can start testing your calls to make sure they all

complete correctly. In order to test redundancy you can shutdown the

gatekeeper or the corresponding T1/E1 controller to test.

Time Saver - By saving the testing until your call routing is complete you

can save time by not interrupting your configuration and then fixing as you

test.

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Codec and Call Admission Control

1. This task was already accomplished earlier when we

created the regions.

2. This task was already accomplished earlier when we

created the regions.

3. This task was completed when we create the locations

earlier.

4. In order to accomplish this task we need to enable

zone bandwidth control on the gatekeeper:

HQ_Rtr(config)#gatekeeper

HQ_Rtr(config-gk)#bandwidth total zone CME 48

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High Availability Features

1. In order to accomplish this task we need to enable

“call-manager-fallback” in the BR1 gateway:

application

global

service alternate DEFAULT

call-manager-fallback

secondary-dialtone 9

max-conferences 4 gain -6

ip source-address 10.1.130.1 port 2000

max-ephones 2

max-dn 10

transfer-pattern .T

voicemail 917029461170

call-forward pattern .T

2. This task was accomplished earlier by setting the BR1

phone 1 device pool to “BR1-NOSRST” which had the SRST

reference disabled.

3. This was completed during the call routing section as

the dial-peers were created with 9 at the beginning.

4. In order to accomplish this we need to add the proper

COR list to the DN of BR1 phone 2 in the incoming

direction:

call-manager-fallback

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cor incoming Phone2 1 2002

5. Most of this task has already been completed during

the phone setup section. We created the AAR groups

with the appropriate prefixes and then assigned the

DN’s to the appropriate AAR group. The last step is to

enable the AAR service via the callmanager service

parameters:

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Media Resources Management

1. In order to accomplish this task we first need to

create the resource in callmanager:

Next we need to configure the SCCP and DSPFARM

settings on the HQ router:

sccp local Loopback0

sccp ccm 10.1.200.21 identifier 2 priority 2 version 7.0

sccp ccm 10.1.200.25 identifier 1 priority 1 version 7.0

sccp

!

sccp ccm group 1

associate ccm 1 priority 1

associate ccm 2 priority 2

associate profile 2 register HQTRANS

associate profile 1 register HQCONF

!

dspfarm profile 2 transcode

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codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

maximum sessions 4

associate application SCCP

!

dspfarm profile 1 conference

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

codec g729r8

codec g729br8

maximum sessions 4

associate application SCCP

!Tip - You can use the “show sccp” and the show “dspfarm profile” command

to verify gateway configuration. You may need to disable and then enable the

SCCP function if the resources aren’t registering.

2. In order to accomplish this task we first need to

create the resource in callmanager:

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Time Saver - This is another great instance of where copy and paste can

help you as the configuration will be very similar to the HQ router.

sccp local Loopback0

sccp ccm 10.1.200.25 identifier 1 priority 1 version 7.0

sccp ccm 10.1.200.21 identifier 2 priority 2 version 7.0

sccp

!

sccp ccm group 1

associate ccm 1 priority 1

associate ccm 2 priority 2

associate profile 1 register BR1CONF

associate profile 2 register BR1TRANS

!

dspfarm profile 2 transcode

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

codec gsmfr

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maximum sessions 4

associate application SCCP

!

dspfarm profile 1 conference

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

codec g729r8

codec g729br8

maximum sessions 4

associate application SCCP

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3. To accomplish this task we first need to create the

resource in callmanager:

The gateway configuration portion was accomplished

above in the conference bridge section.

4. To accomplish this task we first need to create the

resource in callmanager:

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The gateway configuration portion was accomplished

above in the conference bridge section.

5. In a previous step we create all the media resource

groups and list. Now we can add the proper media

resources to the proper groups. The publisher MOH

server will need to be put into the “PUB_MOH” media

resource group and we will need to make sure that this

is the MOH only server listed in the media resource

list for HQ.

6. We need to perform the same actions for BR1 but with

the subscriber server being put into the “SUB_MOH”

media resource group and making sure that this is the

only MOH server listed in the media resource list for

BR1. In addition we will need to enable the default

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audio source, subscriber MOH server, and the “SUB_MOH”

media resource group for multicasting:

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We need to insure that the “Max Hops” is set to “1” so

that the multicast stream stays local.

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Now we need to enable the G.729 codec through the

Cisco IP Voice Media Streaming App service parameters:

We then need to configure the BR1 gateway to support

multicast and the router to play the MOH file from

flash:

call-manager-fallback

moh music-on-hold.au

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multicast moh 239.1.1.1 port 16388 route 10.1.130.1

ip multicast-routing

interface Vlan130

ip pim sparse-dense-mode

! Tip - To test place a call from HQ to BR1 and then put the call on hold from

both ends. You should receive MOH but hear different music on both sides.

7. In order to accomplish this we first need to create

the MeetMe conference resources in callmanager:

We then need to change the callmanager service

parameter to allow for 10 participants per conference:

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8. This is accomplished through the same callmanager

service parameter section as the task above by setting

the “Drop Ad Hoc Conference” parameter correctly.

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QoS Features

1. We can accomplish this with the following

configuration.

On the HQ gateway:

interface Serial0/2/0.1 point-to-point

bandwidth 384

frame-relay interface-dlci 201

auto qos voip trust fr-atm

On the BR1 Gateway:

interface Serial0/2/0.1 point-to-point

bandwidth 384

frame-relay interface-dlci 101 ppp

auto qos voip trust fr-atm

The on both gateways modify the auto generated policy

map per the requirements:

policy-map AutoQoS-Policy-Trust

class AutoQoS-VoIP-RTP-Trust

priority percent 33

class AutoQoS-VoIP-Control-Trust

bandwidth percent 5

class class-default

fair-queue

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2. We can accomplish this with the following

configuration.

On the HQ gateway:

interface Serial0/2/0.2 point-to-point

bandwidth 1200

frame-relay interface-dlci 202

auto qos voip trust

On the BR2 gateway:

interface Serial0/2/0.1 point-to-point

bandwidth 1200

frame-relay interface-dlci 102

auto qos voip trust

Then on both gateways modify the auto generated policy

map per the requirements:

policy-map AutoQoS-Policy-Trust2

class AutoQoS-VoIP-RTP-Trust

priority 360

class AutoQoS-VoIP-Control-Trust

bandwidth 60

class class-default

fair-queue

3. This was accomplished earlier under the two dial-peers

pointing traffic back to the callmanagers.

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Application and End Users

In order to move forward past this section we need to

define all the users. Outlined below in tables are the

users that need to be created and the settings for those

users.

!Tip - Pay close attention to the associated devices and rights of each user.

Before you start to add user you will need to create a “Standard AXL User”

group and assign the “Standard AXL API Access” role to that group.

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!""#$%&'$()*+,-.*

+,-.*/0* /)1(.2&'$()*

!"#$%&&

'(()#*+"#,-&./01&2-3,14+"#,-&

510/0-*0&$1,6(& 7"+-8+18&510/0-*0&91,6(&

'**0("&510/0-*0&76:/*1#("#,-& ;&

'**0("&<6"=,3=8#+),9&>?@?>& ;&

'**0("&.-/,)#*#"08&;,"#3#*+"#,-& ;&

'**0("&>0()+*0/&A0+801& ;&

B0C#*0&2-3,14+"#,-&

!,-"1,))08&B0C#*0/& 7?5DDEF!GHBIB!!&7?5DDIDJK!HGLFK&7?5DDIDJK!HB'IL&7?5DDIDJK!IB!@J&

!'5@&2-3,14+"#,-&

'//,*#+"08&!'5@&51,3#)0/& &

5014#//#,-/&2-3,14+"#,-&

$1,6(/& 7"+-8+18&!M2&')),%&!,-"1,)&,3&'))&B0C#*0/&7"+-8+18&!M2&?-+:)08&

>,)0/& 7"+-8+18&!M2&')),%&!,-"1,)&,3&'))&B0C#*0/&7"+-8+18&!M2&?-+:)08&

&&

25!!&&

'(()#*+"#,-&./01&2-3,14+"#,-&

510/0-*0&$1,6(& 7"+-8+18&510/0-*0&91,6(&

'**0("&510/0-*0&76:/*1#("#,-& ;&

'**0("&<6"=,3=8#+),9&>?@?>& ;&

'**0("&.-/,)#*#"08&;,"#3#*+"#,-& ;&

'**0("&>0()+*0/&A0+801& ;&

B0C#*0&2-3,14+"#,-&

!,-"1,))08&B0C#*0/& &

!'5@&2-3,14+"#,-&

'//,*#+"08&!'5@&51,3#)0/& &

5014#//#,-/&2-3,14+"#,-&

$1,6(/& 7"+-8+18&'NO&./01&

>,)0/& 7"+-8+18&'NO&'52&'**0//&

&&

.-#"P&&

'(()#*+"#,-&./01&2-3,14+"#,-&

510/0-*0&$1,6(& 7"+-8+18&510/0-*0&91,6(&

'**0("&510/0-*0&76:/*1#("#,-& ;&

'**0("&<6"=,3=8#+),9&>?@?>& ;&

'**0("&.-/,)#*#"08&;,"#3#*+"#,-& ;&

'**0("&>0()+*0/&A0+801& ;&

B0C#*0&2-3,14+"#,-&

!,-"1,))08&B0C#*0/& &

!'5@&2-3,14+"#,-&

'//,*#+"08&!'5@&51,3#)0/& &

5014#//#,-/&2-3,14+"#,-&

$1,6(/& 7"+-8+18&'NO&./01&

>,)0/& 7"+-8+18&'NO&'52&'**0//&

&

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The next list of Application users is created during the

setup of the contact center server and therefore will not

need to be create:

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!""#$%&'$()*+,-.*

+,-.*/0* /)1(.2&'$()*

!

"#$%&'('!!

)%%*&+$#&,-!./01!2-3,14$#&,-!

510/0-+0!61,7%! 8#$-9$19!510/0-+0!:1,7%!

)++0%#!510/0-+0!87;/+1&%#&,-! <!

)++0%#!=7#>,3>9&$*,:!?@A@?! <!

)++0%#!.-/,*&+&#09!<,#&3&+$#&,-! <!

)++0%#!?0%*$+0/!B0$901! <!

C0D&+0!2-3,14$#&,-!

E,-#1,**09!C0D&+0/! ))('F'G!))('F'H!))('F'I!))('F'J!))('F'K!))!2EC!

E)5A!2-3,14$#&,-!

)//,+&$#09!E)5A!51,3&*0/! !

5014&//&,-/!2-3,14$#&,-!

61,7%/! 8#$-9$19!EL2!@-$;*09!

?,*0/! 8#$-9$19!EL2!@-$;*09!

!!

14"#$%&'!!

)%%*&+$#&,-!./01!2-3,14$#&,-!

510/0-+0!61,7%! 8#$-9$19!510/0-+0!:1,7%!

)++0%#!510/0-+0!87;/+1&%#&,-! <!

)++0%#!=7#>,3>9&$*,:!?@A@?! <!

)++0%#!.-/,*&+&#09!<,#&3&+$#&,-! <!

)++0%#!?0%*$+0/!B0$901! <!

C0D&+0!2-3,14$#&,-!

E,-#1,**09!C0D&+0/! 8@5FF'HEIMCNCEE!8@5FFNFKOEMIJHO!8@5FFNFKOEMC)NJ!8@5FFNFKOENCEAK!

E)5A!2-3,14$#&,-!

)//,+&$#09!E)5A!51,3&*0/! !

5014&//&,-/!2-3,14$#&,-!

61,7%/! 8#$-9$19!EL2!@-$;*09!

?,*0/! 8#$-9$19!EL2!@-$;*09!

!!

!

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The following End Users will also need to be created with

the correct settings from the table:

!"#$%&'($)*+"#'"&'#,$

%&'($-.$ /'0'12+"'$

3456'($

*+"7(+00'#$

.'89:'&$

*+"7(+00'#$

;(+<90'&$

=&&+:9>7'#$

!?7'"&9+"&$

@(+41&$ A+0'&$

!"#$ %&&#$ '()&&*&+,-*.-/+$ $$ %&&#$01$)23!"#341567189$%&&#$3$4)--$

'581:87:$--;$(1:$<=67=$'581:87:$-24$(18>96:$

'581:87:$--;$(1:$<=67=$'581:87:$--;<'("$?:@010=57850A1$'581:87:$-24$(18>96:$

!"%$ %&&%$ '()&&#B-C%.*.--$ $$ %&&%$01$)23!"#341567189$

'581:87:$--;$(1:$<=67=$'581:87:$-24$(18>96:$

'581:87:$--;$(1:$<=67=$'581:87:$--;<'("$?:@010=57850A1$'581:87:$-24$(18>96:$

DE#$ #&&#$ '()&&*&+,-%CFB,$<)-DE#$

$$ #&&#$01$)23DE341567189$#&&#$3$4)--$

'581:87:$--;$(1:$<=67=$'581:87:$-24$(18>96:$

'581:87:$--;$(1:$<=67=$'581:87:$--;<'("$?:@010=57850A1$'581:87:$-24$(18>96:$

DE%$ #&&%$ '()&&*&+,-%.?*F$ $$ #&&%$01$)23DE341567189$

'581:87:$--;$(1:$<=67=$'581:87:$-24$(18>96:$

'581:87:$--;$(1:$<=67=$'581:87:$--;<'("$?:@010=57850A1$'581:87:$-24$(18>96:$

G7=8:@01$ $$ $$ $$ H$IA16$J$ $$ $$

KAL1$ #&&C$ $$ MAL1$ #&&C$01$)23DE341567189$

'581:87:$--;$(1:$<=67=$'581:87:$-24$(18>96:$

'581:87:$--;$(1:$<=67=$'581:87:$--;<'("$?:@010=57850A1$'581:87:$-24$(18>96:$

!

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Voicemail Integrations

1. In order to accomplish this we first need to perform

all the required setup in callmanager for voicemail.

We start first with creating the voicemail ports:

Time Saver - You can create the ports manually or through the wizard.

Obviously the wizard is designed to save time as it creates the line group

for you will all the voicemail ports as members.

Below is the line group configuration:

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Next you will need to add this line group to a hunt list:

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Now we need to create the voicemail pilot (Hunt Pilot)

and point it to the correct hunt list:

We can now create the MWI on and MWI off extensions:

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We can now set the default voicemail pilot and

profile:

!Tip - You will see that we create a profile named strip. This will be used later

in task 5 in order to strip the “*” out of the extension before it rolls to

voicemail. The settings of this profile are shown below.

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Now in order to start the integration with Unity

Connections we will first need to create a port group:

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Next will need to define the callmanager servers Unity

will use through the Edit->Servers page under the port

group:

During the port group creation we were asked how many

ports and the device name prefix. This created the

ports for us automatically:

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Now we can finish the integration by settings the

parameters for the phone system to include the AXL

servers for importing users from callmanager:

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! Tip - This is also where the loop detection is enabled which is required in the

next task.

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Now we can define the AXL servers through the Edit->

Cisco Unified Communications Manager AXl Servers under

the phone system:

!Note - You will see the application user “Unity” which we create previously.

This user must be in the “Standard AXL User Group” that we created in order

to import users from callmanager.

2. In Unity Connections loop prevention is on by default.

However, you should be aware that there are two ways to

accomplish loop prevention. One is by extension and the

other is through DTMF tone based recognition.

3. In order to accomplish this task you must first modify

the default voicemail user template to change the default

password. You can then import all the users into Unity:

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You can now go to the import users page an import all

users:

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When finished your users page should look like the

following:

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! Tip - If the primary extension of the user is not set then the user will not show

up for import.

4. In order to configure the auto attendant we will first

need to create the appropriate CTI route point in

callmanager and forward all calls to voicemail for this

CTI route point:

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Time Saver -You can see two CTI route points listed as the second one will

be used to record the greeting for the call handler in this task.

We can setup the Greetings Administrator access for HQ

phone 1 user through the call routing section:

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Once this is complete we can now setup the call

handler for the auto attendant:

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Next we can setup the caller input options as required

in the task:

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We now need to enable and setup the closed extension

to take a message:

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We also need to modify the standard greeting so that a

message is not taken:

We need to also assign the call handler owner so that

the HQ phone 1 user can access the call handler and

change the greeting via the Greetings Administrator:

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5. Part of this task has already been configured during the

phone setup section and callmanager voicemail section.

What we need to double check is that the voicemail

profile named “STRIP” is assigned to the BR1 phone 2

extension of “*2002”:

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6. In order to accomplish this task we first need to define

access to the CUE module, the sip-ua for MWI, and then

dial-peers for the correct number(s) that need to be

forwarded to CUE:

interface Service-Engine1/0

ip unnumbered FastEthernet0/0.230

service-module ip address 10.1.230.2 255.255.255.0

service-module ip default-gateway 10.1.230.1

ip route 10.1.230.2 255.255.255.255 Service-Engine1/0

sip-ua

mwi-server ipv4:10.1.230.2 expires 3600 port 5060 transport udp

dial-peer voice 3180 voip

destination-pattern 3180

b2bua

session protocol sipv2

session target ipv4:10.1.230.2

dtmf-relay rtp-nte

codec g711ulaw

no vad

!

dial-peer voice 3000 voip

destination-pattern 3000

b2bua

session protocol sipv2

session target ipv4:10.1.230.2

dtmf-relay rtp-nte

codec g711ulaw

no vad

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Time Saver - In order to save time we can go ahed and create the dial-peer

need for task 8.

You can now start the CUE initialization process by

session into the CUE module from the BR2 gateway with

the following command:

BR2#service-module service-engine 1/0 session

Once you have completed the setup the CUE module will

reboot and is ready for configuration via the web

interface or via command line. You will now need to

set the DTMF relay for the CUE module to match the CME

SIP setup:

ccn subsystem sip

gateway address "10.1.230.1"

dtmf-relay rtp-nte

mwi sip outcall sub-notify

end subsystem

You can test access by pressing the voicemail button

on any BR2 phone.

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7. You can accomplish this via the web interface or via the

command line. We will show through the command line:

username br2phone3 create username BR2Phone2 create username BR2Phone1 create username admin create

username BR2Phone2 phonenumber "3002"

username BR2Phone1 phonenumber "3001"

username BR2Phone2 phonenumberE164 "442076303002"

username BR2Phone1 phonenumberE164 "442076303001"

voicemail mailbox owner "BR2Phone1" size 21176 description "BR2Phone1 mailbox" end mailbox voicemail mailbox owner "BR2Phone2" size 21176 description "BR2Phone2 mailbox" end mailbox

! Tip - You can test access and MWI by leaving a message and making sure the

MWI is working for each mailbox.

8. Part of this task was completed in task 6 and now we just

need to define the auto attendant in CUE:

ccn trigger sip phonenumber 3000

application "autoattendant"

enabled

maxsessions 8

end trigger

9. The was already accomplished earlier during the call

routing section. You will just need to test!

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IPCC Express Voice Application

1. In order to accomplish this task you will need to

complete the normal IPCC Express setup. Once this is

done you will need to create a Call Control Group

based on the task requirements:

!Tip - If required you can click the “Show More...” button and set the partition,

calling search space, AAR group, and other settings that may be required. In

this case we would probably want to set things like the device pool and

location.

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2. To accomplish this task you will need to create a new

application and trigger using the default auto

attendant script:

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3. First we need to create the application and trigger:

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We now can modify the resources and assign the skills

required for task 5:

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! Tip - If you don’t see any resources in IPCC Express then you probably

haven’t assigned the users an IPCC extension in callmanager.

We now need to create the contact services queue with

the same name we referenced under the application:

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We now need to create the IP Agent Phone Service:

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We can then assign the IP phone service to the proper

phones:

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You should now be able to login to the IP Agent Phone

service and test calls into the queue.

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4. In order to accomplish this we can modify the default

script to place the call on hold and then off hold

during the queuing process:

5. This was accomplished in task 3 by assigning the users

skills and then modifying the queue to route based on

“Most Skilled”.

6. This is shown in the script example provided under

task 4.

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! Tip - You should get to know the “Get Reporting Statistics” function in IPCC

Express. The function will allow you to create checks for just about any task.

CallManager Voice Application

1. Part of this task was already completed when we created

the user in callmanager. We now need to create the IP

phone service:

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Next we can create the device profile:

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We need to make sure we subscribe the device profile

to the extension mobility phone service:

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We then need to enable the required phones for extension

mobility and subscribe them to the IP phone service for

extension mobility:

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Presence

1. The first step is to run through the setup for

Presence after the initial install. Once this is

complete we can setup the required SIP trunk in

callmanager and set the SIP publish trunk in the

callmanager service parameters:

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Now set the CUP Publish Trunk in the callmanager

service parameters:

We can now create the CUPC in callmanager:

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Make sure on all the phones and CUPC in callmanager

that the subscribe calling search space is set

accordingly:

Next we need to assign the capabilities in

callmanager:

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Once this is complete we now need to finish the

configuration of the presence server. We can start by

validating the setup via System->CUCM Publisher:

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You can now configure the incoming and outgoing ACL’s:

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We now need to configure the CM SIP Publish trunk

under the Presence->Settings:

We now need to configure the callmanager as a presence

gateway:

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We now need to modify the proxy services settings:

We can then start configuring the CUPC settings:

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Next we need to configure the Unity server as a

voicemail server and then configure the voicemail

profile:

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We can now set the proper profiles for the HQ phone 1

user:

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Next we need to configure the “Deskphone Control

Settings” so that the CUPC client can control the

phone:

! Tip - The phones that need to be controlled by the “CtiGw” user were defined

earlier during the Application User section.

! Tip - You will also want to make sure that the phone being controlled has an

owner assigned and is associated to the end user.

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We can then enable the HQ phone 1 user for desktop

phone control:

Next you will need to the proxy domain and restart the

presence and proxy services:

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!Tip - You will need to create a host file entry on the machine running CUPC

based on the domain name you select in the presence server parameters.

Example: “presence.ccbootcamp.com” pointing to 10.1.200.23.

2. We need to first enable Unified Client access within

Unity Connections:

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Next we need to setup the Unity account user name and

password in the CUPC client and check that all

services are up and running:

!Note - You can switch between modes and check services for each setup. Keep

in mind that the service for the mode your not in will show as down or not

configured.

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Chapter 3 - Lab 3

Brief Lab Overview:

This lab is designed to test your CME knowledge. The focus

is CME SCCP features.

Basic Campus Design:

1. Configure VLANS and IP networks in all three sites

according to the following two tables:

!

VLAN HQ BR1 BR2

Servers 1 N/A N/A

Data 20 120 220

Voice 30 130 230

!

Network HQ BR1 BR2

Servers 10.1.200.0/24 N/A N/A

Data 10.YY.20.0/24 10.YY.120.0/24 10.YY.220.0/24

Voice 10.YY.30.0/24 10.YY.130.0/24 10.YY.230.0/24

2. Make sure you set all voice traffic to use the voice

VLAN for all sites.

3. Configure phones on HQ site to have IP addresses

120-130. Use IOS DHCP on the HQ router to accomplish

this task.

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4. Configure BR1 phones to have IP addresses of 130–140

using the IOS DHCP on the BR1 Router.

5. Configure BR2 phones to have IP addresses 210–220

using the IOS DHCP on the BR2 router.

6. Configure HQ router as an NTP master clock with

stratum 2 for local time zone in Los Angeles and

configure BR1 and BR2 to synchronize their clocks to

HQ. BR1 is in New York time zone and BR2 is in India.

CallManager and CallManager Express:

1. Register all phones according to the dial plan table

using SCCP loads. Make sure the phones will display

calling names in every scenario presented (internal

and PSTN):

2. Configure directory number 3003 as a second line on

both phones in BR2. When a call comes into 3003 it

should ring on both phones and be answered by the

first user to respond.

3. Make sure you that when the initiator of a conference

call within the BR2 site drops that the conference

call drops.

4. Make sure all phones display the same general

information aside from the line specific

configuration; this refers to HQ and BR1.

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5. Allow BR2 phone 1 to pickup calls ringing on BR2

phone 2 and vice versa. This should work for internal

and external calls.

6. Both phone 1 and phone 2 are members of VIP sales

group and are required to take calls dialed to

442076303111. The calls are expected to ring the

phones in sequential order. Both phones should ring

for 5 seconds and if the call is not answered it

should be forwarded to DN 3000.

7. Set BR2 phone 1 for extension mobility with an auto-

logout enabled. User BR2 Phone 1 should be able to

login to BR2 phone 1 and have all their settings

enabled.

8. Enable both phones at site BR2 for VoiceView Express.

9. Configure BR2 phone 2 so that when the phone is

called it rings our to 6305000 after 5 seconds and

then forwards to voicemail after 10 seconds.

Voice Gateways and Signaling:

1. Configure the HQ router as a IOS MGCP gateway using T1

PRI and NI as the line parameters.

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2. Configure BR1 router as a IOS H.323 gateway using T1

PRI and NI as the line parameters.

3. Configure BR2 as E1 PRI and make sure all BR2 numbers

are reachable.

4. Configure HQ router as a gatekeeper with the following

details:

Local zone name: voiceie

Domain name: ccbootcamp.com

Register CallManager and CME to the gatekeeper.

Register the CallManager with a tech-prefix of “1#”

and a zone called “CM”. Register CME with a tech-

prefix of “1#” and to a zone called “CME”.

Call Routing:

1. In every site, phone 1 should be allowed to dial

international and below numbers, phone 2 should only

be allowed to call long distance and below.

2. Configure the following dialing options for users in

HQ:

7+4 digits Calls to BR2

911 and 9911 Emergency

9+7 digits

The first is in the range of

2 to 9

Local

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9+1+10 digits

The first and the forth are

in the range of 2 to 9.

Long Distance

9+011+any number of digits

Please allow users to dial

the hash sign at the end of

the number in order to have

faster response

International

• Local calls from HQ should use local HQ gateway

and BR1 gateway as backup.

• Long distance calls from HQ should use local HQ

gateway as first choice and then BR1 gateway.

• Calls to BR1 PSTN numbers should be routed out

BR1 gateway (Toll Bypass), with fallback to local

HQ gateway.

• International calls to BR2 numbers, using the 7+4

digits pattern, should be routed via the

gatekeeper as VoIP and use the local HQ gateway

as backup.

• International calls to BR2 local numbers should

be routed via the gatekeeper and then via the

local HQ gateway.

• All other International calls should route out

the HQ gateway.

3. Configure the following dialing options for users in

BR1:

7+4 digits Calls to BR2

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911 and 9911 Emergency

9+7 digits

The first is in the range of 2

to 9

Local

9+1+10 digits

The first and the forth are in

the range of 2 to 9.

Long Distance

9+011+any number of digits

Please allow users to dial the

hash sign at the end of the

number in order to have faster

response

International

• Local calls from BR1 should use the local BR1 IOS

gateway and then the HQ gateway as a backup.

• Long distance calls from BR1 should use the BR1

gateway as a first choice and then the HQ gateway

as a backup.

• Calls to HQ PSTN numbers (Area Code+Prefix)

should be routed out the HQ gateway (Toll Bypass)

with fallback to the local BR1 gateway.

• International calls to BR2 numbers, using the 7+4

digits pattern, should be routed via the

gatekeeper as VoIP and use the BR1 gateway as

backup.

• International calls to BR2 local numbers should

be routed via the gatekeeper and then via the

local HQ gateway.

• All other International calls should route out

the BR1 gateway.

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4. Configure the following dialing options for users in

BR2:

7+4 digits Calls to HQ and BR1

999 Emergency

9+7 digits Local

9+0+8 digits Long Distance

9+00+any number of digits

Please allow users to dial

the hash sign at the end of

the number in order to have

faster response

International

• International calls to HQ and BR1 should use the

gatekeeper first and the local PRI as a backup.

Codec and Call Admission Control:

1. All calls within a site should use the G.711 codec.

2. All calls between sites should use the G.729 codec.

3. Allow two concurrent calls to and from BR1.

4. Allow three concurrent calls from HQ and BR1 to BR2.

High Availability Features:

1. Configure BR1 as an SRST router.

2. Make sure you only allow BR1 phone 2 to register via

SRST during a failure.

3. Preserve all PSTN dialing with using 9 as the first

digit in the string.

4. When in SRST mode, the BR1 phone 2 is not allowed to

dial international calls.

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5. Allow HQ and BR1 phones to call each other in times

when more then 2 concurrent calls are in process

between the sites.

Media Resources Management:

1. Configure conferencing for the HQ site to use hardware

based resources in the HQ gateway first then hardware

resources in the BR1 gateway.

2. Configure conferencing for the BR1 site to use

hardware based resources in the BR1 gateway first then

hardware resources in the HQ gateway.

3. Configure transcoding for the HQ site to use hardware

based resources in the HQ gateway first then hardware

resources in the BR1 gateway.

4. Configure transcoding for the BR1 site to use hardware

based resources in the BR1 gateway first then hardware

resources in the HQ gateway.

5. Configure MOH for the HQ site based on unicast and

streamed from the publisher CallManager.

6. Configure multicast MOH for BR1, based on G729 codec

from the subscriber, and streamed from the BR1

gateway.

7. Allow 5 simultaneous meet me conferences for the HQ

users, in the number range 1201-1205. Every meet me

conference should allow a maximum of ten participants.

8. Make sure that when the initiator of an Ad-Hoc

conference hangs up, the conference is allowed to

continue as long as an internal user is still

connected.

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QoS Features:

7. Configure the frame relay link between HQ and BR1 as

if it is 384k in bandwidth, make sure you enable MLP

LFI on it.

• VoIP signaling should have 5% of the line as

guarantied bandwidth.

• VoIP barrier should have 33% of the line as priority

bandwidth.

• All other traffic should be weighted fair queued.

8. Configure the frame relay link between HQ and BR2 as

if it is 1200k in bandwidth.

• VoIP signaling should have 60k of the line reserved

bandwidth.

• VoIP barrier should have 360k of the line priority

bandwidth.

• All other traffic should be weighted fair queued.

9. The BR1 gateway should send the signaling and RTP

traffic marked as CS3 and EF.

Voicemail Integrations:

1. Configure voicemail for HQ and BR1 based on the Unity

Connections server using the following parameters:

• Pilot 1170, port 1171-1174, MWI on 1180, MWI off

1181.

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2. Make sure you prevent Unity Connections from calling

to itself and creating call loops which consume the

available ports.

3. Configure a voicemail box for phones 1 and 2 in HQ and

BR1 with a phone password of 147369 for all users.

4. Configure an auto attendant for HQ with DN 1000. It

should have a customized greeting that allows dialing

an extension and walking through the users directory

by pressing 1. When users press 0 they should be

forwarded to extension 1001. During closed times,

assume normal weekday schedule, it should take a

message and store it in the HQ phone 1 mailbox.

5. Create a second line on BR1 phone 2 with extension

*2002. A call coming to the primary line should roll

to this line if the primary is busy. The user should

be able to use this line for voicemail access. Do not

use any alternate extensions.

6. Configure BR2 CME to use the local Unity Express as a

voicemail, using the following parameters:

• Pilot 3180, MWI on 8001, MWI off 8002.

7. Create voicemail for BR2 phone 1 and 2. Use a blank

phone password for all users.

8. Create an auto attendant for BR2 that will work when

DN 3000 is dialed.

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9. Make sure that calls from HQ can route via gatekeeper

and hit the auto attendant in BR2.

IPCC Express Voice Application:

1. Configure the IPCC on the Contact Center server to

work with the CallManager using the following

parameters (all passwords are “cisco” and all pin’s

are “12345”):

Jtapi username: jtapi

Resource manager user: rmjtapi

CTI ports: 1015 – 1019.

2. Configure the auto attendant script to answer when a

call arrives at 1020.

3. Configure an ICD queue that will be triggered when

dialing extension 1021. HQ phone 1 should be used as

an IP phone agent for user “HQ1” with the password

“cisco”. BR1 phone 1 should be used as a IP phone

agent with user “BR1” with a password “cisco”.

4. Callers should hear music on hold while delayed in

queue.

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5. Calls should be routed via the queue based on a skill

named “Sales”. HQ phone 1 should be skilled higher

than BR1 phone 1.

6. Before a call is routed to the queue a check should be

performed to see if any agents are logged into the

queue. If no agents are logged in then the call should

be routed to HQ phone 2.

CallManager Voice Application:

1. Create extension mobility for user “john” with a

password of “cisco”. Which will allow him to use

extension 1007 on HQ and BR1 phone 2 in each site.

Make sure that the extension mobility service is only

available on phone 2 in HQ and BR1.

Presence:

1. Integrate the Presence server with CallManager. Make

sure the HQ Phone 1 user can login via CUPC and

control the desktop phone. HQ Phone 1 user should also

be allowed to switch to softphone mode.

2. Make sure HQ Phone 1 can see the voicemail in their

voicemail box through the CUPC client.

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Solutions:

Basic Campus Design

1. Start with “show vlan” and make sure you have all

VLANS defined in all devices. If not create them and

then add the IP interfaces to each device with the

correct IP subnet from the table provided:

Time Saver - In order to save time you should go ahead and configure the

layer 2 QoS in the HQ switch.

HQ:

HQ-3750#

HQ-3750#config t

Enter configuration commands, one per line. End with CNTL/Z.

HQ-3750(config)#vlan 20!! !

HQ-3750(config-vlan)#name Data

HQ-3750(config-vlan)#exit

HQ-3750(config)#vlan 30!! !

HQ-3750(config-vlan)#name Voice

HQ-3750(config-vlan)#exit

HQ-3750(config)#

HQ-3750(config)#interface range FastEthernet 1/0/6 - 7

HQ-3750(config-if-range)#switchport voice vlan 30

HQ-3750(config-if-range)#switchport access vlan 20

HQ-3750(config-if-range)#switchport trunk encapsulation dot1q

HQ-3750(config-if-range)#switchport mode trunk

HQ-3750(config-if-range)#auto qos voip cisco-phone

HQ-3750(config)#interface range FastEthernet 1/0/1 - 5

HQ-3750(config-if-range)#switchport host

HQ-3750(config-if-range)#auto qos voip trust

HQ-3750(config)#interface FastEthernet 1/0/24

HQ-3750(config-if)#auto qos voip trust

HQ-3750(config)#interface FastEthernet0/0.20

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HQ-3750(config-if)#encapsulation dot1Q 20

HQ-3750(config-if)#ip address 10.1.20.1 255.255.255.0

HQ-3750(config)#exit

HQ-3750(config)#interface FastEthernet0/0.30

HQ-3750(config-if)#encapsulation dot1Q 30

HQ-3750(config-if)#ip address 10.1.30.1 255.255.255.0

! Tip - Don’t forget to add the new networks into the routing protocol.

HQ-3750(config)#router ospf 1

HQ-3750(config-router)#network 10.1.20.0 0.0.0.255 area 0

HQ-3750(config-router)#network 10.1.30.0 0.0.0.255 area 0

BR1:

BR1#vlan database

BR1(vlan)#vlan 120 name Data state active

VLAN 120 added:

Name: Data

State ACTIVE

BR1(vlan)#vlan 130 name Voice state active

VLAN 130 added:

Name: Voice

State ACTIVE

BR1(vlan)#exit

APPLY completed.

Exiting....

! Tip - The “vlan database” command has been deprecated so you can now use

the “vlan” command under global configuration mode as well.

BR1#config t

Enter configuration commands, one per line. End with CNTL/Z.

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BR1(config)#int vlan 120

BR1(config-if)#ip address 10.1.120.1 255.255.255.0

BR1(config-if)#description Data

BR1(config-if)#int vlan 130

BR1(config-if)#ip address 10.1.130.1 255.255.255.0

BR1(config-if)#description Voice

BR1(config-if)#exit

BR1(config)#interface range FastEthernet 0/3/0 - 1

BR1(config-if-range)#switchport voice vlan 130

BR1(config-if-range)#switchport access vlan 120

BR1(config-if-range)#no shut

! Tip - Don’t forget to add the new networks into the routing protocol.

BR1(config)#router ospf 1

BR1(config-router)#network 10.1.120.0 0.0.0.255 area 0

BR1(config-router)#network 10.1.130.0 0.0.0.255 area 0

BR2:

BR2#vlan database

BR2(vlan)#vlan 220 name Data state active

VLAN 220 added:

Name: Data

State ACTIVE

BR2(vlan)#vlan 230 name Voice state active

VLAN 230 added:

Name: Voice

State ACTIVE

BR2(vlan)#exit

APPLY completed.

Exiting....

BR2#config t

Enter configuration commands, one per line. End with CNTL/Z.

BR2(config)#int vlan 220

BR2(config-if)#ip address 10.1.220.1 255.255.255.0

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BR2(config-if)#description Data

BR2(config-if)#int vlan 230

BR2(config-if)#ip address 10.1.230.1 255.255.255.0

BR2(config-if)#description Voice

BR2(config-if)#exit

BR2(config)#interface range FastEthernet 0/1/0 - 1

BR2(config-if-range)#switchport voice vlan 230

BR2(config-if-range)#switchport access vlan 220

BR2(config-if-range)#no shut

! Tip - Don’t forget to add the new networks into the routing protocol.

BR2(config)#router ospf 1

BR2(config-router)#network 10.1.220.0 0.0.0.255 area 0

BR2(config-router)#network 10.1.230.0 0.0.0.255 area 0

2. Validate your configurations with the “show

vlan” ( or “show current” under the vlan database)

and “show cdp neighbors” commands.

!

HQ-3750#show vlan

VLAN Name Status Ports

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

1 default active ! Fa1/0/1, Fa1/0/2, Fa1/0/3, Fa1/0/4,

! ! ! ! ! ! Fa1/0/5, Fa1/0/8, Fa1/0/9, Fa1/0/10,

! ! ! ! ! ! Fa1/0/11, Fa1/0/12, Fa1/0/13, Fa1/0/14,

! ! ! ! ! ! Fa1/0/15, Fa1/0/16, Fa1/0/17, Fa1/0/18,

! ! ! ! ! ! Fa1/0/19, Fa1/0/20, Fa1/0/21, Fa1/0/22,

! ! ! ! ! ! Fa1/0/23, Gi1/0/1, Gi1/0/2

20 Data active

30 Voice active

1002 fddi-default act/unsup

1003 token-ring-default act/unsup

1004 fddinet-default act/unsup

1005 trnet-default act/unsup

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HQ-3750#show cdp neigh

Capability Codes: R - Router, T - Trans Bridge, B - Source Route Bridge

S - Switch, H - Host, I - IGMP, r - Repeater, P - Phone

Device ID Local Intrfce Holdtme Capability Platform Port ID

CCMSUB Fas 1/0/1 152 H none foundeth0

CCMPUB Fas 1/0/1 157 H none foundeth0

HQ_Rtr Fas 1/0/24 140 R S I 2811 Fas 0/0.1

uccx. Fas 1/0/5 172 H Win2000 SeEth 1/1

presence Fas 1/0/4 145 H none foundeth0

SEP003094C27864 Fas 1/0/6 138 H P IP Phone 7Port 1

SEP003094C2DA38 Fas 1/0/7 132 H P IP Phone 7Port 1

SEP0002FDAEFDDF Fas 1/0/8 159 H P IP Phone 7Port 1

Unity Fas 1/0/3 162 H none foundeth0

PSTN-WAN Fas 1/0/22 124 R S I 2811 Fas 0/0

HQ-3750#

!Note - The publisher and subscriber are on the same physical box so they will

show as if they are connected to the same port on the switch. This is normal

and you should configure FastEthernet 1/0/2 as if the subscriber is plugged

into that port.

BR1#vlan database

BR1(vlan)#show current

VLAN ISL Id: 1

Name: default

Media Type: Ethernet

VLAN 802.10 Id: 100001

State: Operational

MTU: 1500

Translational Bridged VLAN: 1002

Translational Bridged VLAN: 1003

VLAN ISL Id: 120

Name: Data

Media Type: Ethernet

VLAN 802.10 Id: 100220

State: Operational

MTU: 1500

VLAN ISL Id: 130

Name: Voice

Media Type: Ethernet

VLAN 802.10 Id: 100230

State: Operational

MTU: 1500

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BR1#show cdp neigh

Capability Codes: R - Router, T - Trans Bridge, B - Source Route Bridge

S - Switch, H - Host, I - IGMP, r - Repeater

Device ID Local Intrfce Holdtme Capability Platform Port ID

HQ_Rtr Ser 0/1/0.1 161 R S I 2811 Ser 0/2/0.1

SEP0016C72D3DCC Fas 0/3/1 146 H IP Phone Port 1

SEP003094C3DCF9 Fas 0/3/0 143 H IP Phone Port 1

BR1#

BR2# show cdp neigh

Capability Codes: R - Router, T - Trans Bridge, B - Source Route Bridge

S - Switch, H - Host, I - IGMP, r - Repeater

Device ID Local Intrfce Holdtme Capability Platform Port ID

HQ_Rtr Ser 0/2/0.1 133 R S I 2811 Ser 0/2/0.2

SEP001F6C7FD6C1 Fas 0/1/1 167 H IP Phone Port 1

SEP001F6C7F722B Fas 0/1/0 167 H IP Phone Port 1

BR2#

BR2#vlan database

BR2(vlan)#show current

VLAN ISL Id: 1

Name: default

Media Type: Ethernet

VLAN 802.10 Id: 100001

State: Operational

MTU: 1500

Translational Bridged VLAN: 1002

Translational Bridged VLAN: 1003

VLAN ISL Id: 220

Name: Data

Media Type: Ethernet

VLAN 802.10 Id: 100220

State: Operational

MTU: 1500

VLAN ISL Id: 230

Name: Voice

Media Type: Ethernet

VLAN 802.10 Id: 100230

State: Operational

MTU: 1500

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3. In order to accomplish this we need to first create

the dhcp excluded address and then define the DHCP

pool:

HQ_Rtr(config)#ip dhcp excluded-address 10.1.30.1 10.1.30.119

HQ_Rtr(config)#ip dhcp excluded-address 10.1.30.131 10.1.30.255

HQ_Rtr(config)#ip dhcp pool IPPHONE

HQ_Rtr(config-config)#network 10.1.30.0 255.255.255.0

HQ_Rtr(config-config)#default-router 10.1.30.1

HQ_Rtr(config-config)#option 150 ip 10.1.200.21 10.1.200.25

Time Saver - In order to save time you should use notepad to copy and

paste configurations which may be the same or need little modification.

4. We can use the same configuration from the last task

but modify for BR1:

BR1(config)#ip dhcp excluded-address 10.1.130.1 10.1.130.129

BR1(config)#ip dhcp excluded-address 10.1.130.141 10.1.130.255

BR1(config)#ip dhcp pool IPPHONE

BR1(config-config)#network 10.1.130.0 255.255.255.0

BR1(config-config)#default-router 10.1.130.1

BR1(config-config)#option 150 ip 10.1.200.21 10.1.200.25

5. Perform the same for BR2:

BR2(config)#ip dhcp excluded-address 10.1.230.1 10.1.230.209

BR2(config)#ip dhcp excluded-address 10.1.230.221 10.1.230.255

BR2(config)#ip dhcp pool IPPHONE

BR2(config-config)#network 10.1.230.0 255.255.255.0

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BR2(config-config)#default-router 10.1.230.1

BR2(config-config)#option 150 ip 10.1.230.1

6. Next configure each router for the appropriate time

zone and NTP settings:

HQ_Rtr(config)#clock timezone PST -8

HQ_Rtr(config)#clock summer-time PDT recurring

HQ_Rtr(config)#ntp master 2

BR1(config)#clock timezone EST -5

BR1(config)#clock summer-time EDT recurring

BR1(config)#ntp server 10.10.32.1

BR2(config)#clock timezone UTC +5 30

BR2(config)#clock summer-time PDT recurring

BR2(config)#ntp server 10.10.32.1

! Tip - Use can use your candidate PC to look up the time zone of most

countries but most likely they will specify for you.

! Tip - Don’t forget to set the time zone settings in CallManager as it may not be

mentioned in the phone setup tasks.

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CallManager and CallManager Express:

1. This tasks requires a lot of configuration as this is

the base line for everything you do in CallManager. We

will setup everything from device pools, locations,

regions, media resource lists and groups, route lists

and route groups, partitions and calling search

spaces.

!

Lets start off with service activation first as

nothing will work if we don’t have services activated:

Make sure you enable services on both servers and all

services except the “Cisco Messaging Interface”.

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!Tip - Enabling all services as stated above will ensure that all needed services

are activated. Unless you are specifically told to only enable needed services

then this strategy will allow you to make sure all the required services are

running.

Next we need to start configuring the required

CallManager settings needed to allow for the

registration of HQ and BR1 phones.

Make sure the proper servers are defined:

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Next create a CallManager group that will allow for

devices to register to the subscriber and then the

publisher:

When finished you should see the following to groups:

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! Note - The requirement for registration order is normally on the first page of

instructions.

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Next we need to create the regions:

We need to make sure the settings between the regions

are set appropriately:

Time Saver - This is where reading through the whole lab will save a lot of

time as we can set the proper region settings and never return to this page.

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Now we need to create the HQ and BR1 Date/Time groups:

! Note - This is also where we could set the date/time format if asked during a

specific task.

Next we need to create the locations:

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Time Saver - This is where reading through the whole lab will save a lot of

time as we can set the proper location settings and never return to this page.

We now need to create the media resource groups and

lists:

!Tip - In reading the lab we know that there is a need to separate the MOH

servers and it is always a good idea to separate hardware from software

resources.

We can then create the needed media resource lists:

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Below is the full list of the media resource lists:

We now need to create the device pools:

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Time Saver - We know from reading the lab that one phone will not be

allowed to register in SRST. We can create the device pool for this function

now in order to save time later.

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Here is the final list of device pools:

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Next we need to check the Enterprise Parameters and

make changes as needed:

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!Tip - Changing the host name to an IP address in the URL settings will prevent

authentication and IP phone service failures. This is highly recommended

whether you have a DNS server or not!

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We can also create the AAR groups now as this will

help to cut down the number of times we need to visit

the phone configuration page:

Once we have both groups created we need to set the

dial prefix according to the dial plan:

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Now we will need to create the the partitions and

calling search spaces that will be needed for the lab:

On the next page is a table of calling search spaces

and the partitions that belong to each calling search

space.

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!"##$%&'()"*+,'(-"+)'

."/)' 0)1+*$-2$3%' 4352)'6"*2$2$3%1'73*'2,$1'!"##$%&'()"*+,'(-"+)'

!""#$%&#''%( !""#$%&#''%( )*+$%&+''%(

!""#$%&#,-*( !""#$%&#,-*( )*+$%&+.&&(

)*+$%&+,/012/34(

)*+$%&+*566$7)'""(

)*+89+,/012/34(

)*+$%:+,/012/34(

)*+$%&+6;<34(

)*+$%&+6=(

)*+$%&+,-*(

!""#$%&#,-*>%-'6( !""#$%&#,-*>%-'6( )*+$%&+.&&(

)*+$%&+,/012/34(

)*+$%&+*566$7)'""(

)*+89+,/012/34(

)*+$%:+,/012/34(

!""#$%&#6=( !""#$%&#6=( )*+$%&+.&&(

)*+$%&+,/012/34(

)*+$%&+*566$7)'""(

)*+89+,/012/34(

)*+$%:+,/012/34(

)*+$%&+6;<34(

)*+$%&+6=(

!""#$%&#65!'6( !""#$%&#65!'6( )*+$%&+.&&(

)*+$%&+,/012/34(

)*+$%&+*566$7)'""(

)*+89+,/012/34(

)*+$%:+,/012/34(

)*+$%&+6;<34(

!""#$%:( !""#$%:( )*+$%&+,/012/34(

)*+$%:+,/012/34(

)*+89+,/012/34(

)*+$%:+*566$7)'""(

!""#89#''%( !""#89#''%( )*+89+''%(

!""#89#,-*( !""#89#,-*( )*+89+.&&(

)*+89+,/012/34(

)*+89+*566$7)'""(

)*+$%&+,/012/34(

)*+$%:+,/012/34(

)*+89+6;<34(

)*+89+6=(

)*+89+,-*(

!""#89#,-*>%-'6( !""#89#,-*>%-'6( )*+89+.&&(

)*+89+,/012/34(

)*+89+*566$7)'""(

)*+$%&+,/012/34(

)*+$%:+,/012/34(

!""#89#6=( !""#89#6=( )*+89+.&&(

)*+89+,/012/34(

)*+89+*566$7)'""(

)*+$%&+,/012/34(

)*+$%:+,/012/34(

)*+89+6;<34(

)*+89+6=(

!""#89#65!'6( !""#89#65!'6( )*+89+.&&(

)*+89+,/012/34(

)*+89+*566$7)'""(

)*+$%&+,/012/34(

)*+$%:+,/012/34(

)*+89+6;<34(

!""#?5,!>@',6( !""#?5,!>@',6( )*+$%&+,/012/34(

)*+89+,/012/34(

(

(

We should now be able to add the phones per the

requirements in the table.

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HQ Phone 1:

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HQ Phone 2:

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BR1 Phone 1:

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BR1 Phone 2:

!Note - In reading the lab we know the BR1 Phone 2 requires another extension

“*2002”. We can create that as part of the initial phone setup which we have

done here or you could wait till that specific task.

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BR1 Phone 2 (Line 2):

! Tip - Most of the phone settings per phone are the same per site except for the

calling search space.

Time Saver - In order to save time you can use auto registration on the

callmanager and then modify each phone to the correct settings.

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We can now start the registration of phones within the

BR2 site. We will first need to define all the

required settings underneath the “telephony-service”

configuration. We can then create the phones and their

associated DN’s:

BR2(config)#telephony-service

BR2(config-telephony)#authentication credential admin cisco

BR2(config-telephony)#max-ephones 2

BR2(config-telephony)#max-dn 7

BR2(config-telephony)#system message Your current options

BR2(config-telephony)#ip source-address 10.1.230.1 port 2000

BR2(config-telephony)#load 7941 term41.default.loads

BR2(config-telephony)#voicemail 3180

BR2(config-telephony)#max-conferences 8 gain -6

BR2(config-telephony)#call-forward pattern .T

BR2(config-telephony)#dn-webedit

BR2(config-telephony)#time-webedit

BR2(config-telephony)#transfer-system full-blind

BR2(config-telephony)#transfer-pattern 9.T

BR2(config-telephony)#secondary-dialtone 9

BR2(config-telephony)#create cnf-files

BR2(config-telephony)#exit

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Now we can configure the phones and their DN’s:

BR2(config)#ephone-dn 1

BR2(config-ephone-dn)#number 3001

BR2(config-ephone-dn)#pickup-group 1

BR2(config-ephone-dn)#description 6303001

BR2(config-ephone-dn)#name BR2 Phone 1

BR2(config-ephone-dn)#call-forward noan 3180 timeout 5

BR2(config)#ephone-dn 2

BR2(config-ephone-dn)#number 3002

BR2(config-ephone-dn)#pickup-group 1

BR2(config-ephone-dn)#description 6303002

BR2(config-ephone-dn)#name BR2 Phone 2

We can also configure the extra DN’s specified for the

shared line on BR2 phone 1 and 2:

BR2(config)#ephone-dn 6 dual-line

BR2(config-ephone-dn)# number 3003

BR2(config-ephone-dn)# pickup-group 1

BR2(config-ephone-dn)# description 6303003

BR2(config)#ephone-dn 7 dual-line

BR2(config-ephone-dn)# number 3004

BR2(config-ephone-dn)# pickup-group 1

We can now configure the phones knowing that phone 1

will be using extension mobility:

BR2(config)#ephone 1

BR2(config-ephone)#conference drop-mode creator

BR2(config-ephone)#mac-address 001F.6C7F.722B

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BR2(config-ephone)#type 7941GE

BR2(config-ephone)#logout-profile 1

BR2(config)#ephone 2

BR2(config-ephone)#conference drop-mode creator

BR2(config-ephone)#mac-address 001F.6C7F.D6C1

BR2(config-ephone)#username "BR2Phone2"

BR2(config-ephone)#type 7941GE

BR2(config-ephone)#button 1:2 2o6,7

Next we need to allow the SCCP firmware to be

downloaded from the router flash:

BR2(config)#tftp-server flash:apps41.8-3-2-27.sbn BR2(config)#tftp-server flash:cnu41.8-3-2-27.sbn BR2(config)#tftp-server flash:cvm41sccp.8-3-2-27.sbn BR2(config)#tftp-server flash:dsp41.8-3-2-27.sbn BR2(config)#tftp-server flash:jar41sccp.8-3-2-27.sbn

!Tip - If you like to use the CME and CUE web interface the the following

commands should be entered at this stage to prepare you to use the web

interface.

BR2(config)#ip http server

BR2(config)#no ip http secure-server

BR2(config)#ip http path flash:

BR2(config)#telephony-service

BR2(config-telephony)#web admin system name admin password cisco

BR2(config-telephony)#url services http://10.1.230.2/voiceview/common/

login.do (Inserted by default in most cases)

BR2(config-telephony)#url authentication http://10.1.230.1/CCMCIP/

authenticate.asp (Inserted by default in most cases)

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2. This was already completed in the above solution with the

ephone-dn 6 and 7 configuration and the addition of this

button with overlay to phone 2 ephone. Adding this to

phone 1 will come later during the extension mobility

task.

3. This was already completed in the above solution with the

“conference drop-mode creator” command under the ephone

configuration for phone 2. Adding this to phone 1 will

come later during the extension mobility task.

4. This was accomplished in task 1 by setting the alerting

name, caller id, external phone number mask, and date/

time group format the same in CallManager.

5. This was accomplished in task 1 by setting setting the

“pickup-group 1” command under all the ephone-dn’s.

6. This is accomplished with the following configuration:

BR2(config)#ephone-hunt 1 sequential

BR2(config-ephone-hunt)#pilot 3111

BR2(config-ephone-hunt)#list 3001, 3003

BR2(config-ephone-hunt)#final 3000

BR2(config-ephone-hunt)#timeout 5, 5

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7. This is accomplished through creating a logout profile

and a user profile for phone 1:

BR2(config)#voice logout-profile 1

BR2(config-voice-profile)#pin 12345

BR2(config-voice-profile)#user br2phone3 password cisco

BR2(config-voice-profile)#number 3003,3004 type overlay

BR2(config)#voice user-profile 1

BR2(config-voice-profile)#pin 12345

BR2(config-voice-profile)#user BR2Phone1 password cisco

BR2(config-voice-profile)#number 3001 type normal

BR2(config-voice-profile)#number 3003,3004 type overlay

8. In this version of CUE and CME voice view is enabled by

default. One step that needs to take place is adding the

CME authentication URL as the backup URL for the voice

view service in CUE:

CUE(config)#service phone-authentication CUE(config-service)#fallback-url http://10.1.230.1/CCMCIP/

authenticate.asp CUE(config-service)#end phone-authentication

9. The is simple to accomplish by adding the following under

the ephone-dn:

BR2(config)#ephone-dn 2

BR2(config-ephone-dn)#mobility

BR2(config-ephone-dn)#snr 96305000 delay 5 timeout 10 cfwd-noan 3180

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!Note - Though the task asks us to allow the number to be reach by the full

E164 number we can perform this action later when setting up the call routing

as all calls will need to be striped to four digits.

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Voice Gateways and Signaling

1. In order to enable the HQ router as an MGCP gateway we

need the following configuration on the HQ router:

HQ_Rtr(config)#network-clock-participate wic 0

HQ_Rtr(config)#isdn switch-type primary-ni

HQ_Rtr(config)#voice-card 0

HQ_Rtr(config-voicecard)#dspfarm

HQ_Rtr(config-voicecard)#dsp services dspfarm

HQ_Rtr(config-voicecard)#exit

HQ_Rtr(config)#controller T1 0/0/0

HQ_Rtr(config-controller)#cablelength long 0db

HQ_Rtr(config-controller)#pri-group timeslots 1-3,24 service mgcp

HQ_Rtr(config-controller)#exit

HQ_Rtr(config)#interface Serial0/0/0:23

HQ_Rtr(config-if)# isdn bind-l3 ccm-manager

HQ_Rtr(config-if)#exit

HQ_Rtr(config)#ccm-manager redundant-host 10.1.200.21

HQ_Rtr(config)#ccm-manager mgcp

HQ_Rtr(config)#mgcp

HQ_Rtr(config)#mgcp call-agent 10.1.200.25 service-type mgcp version 0.1

HQ_Rtr(config)#mgcp fax t38 ecm

HQ_Rtr(config)#mgcp

! Note - In IOS release 12.4 the dial-peer no longer needs to be created with the

service command.

Time Saver - If you have multiple MGCP or H.323 gateways then use

notepad to copy and paste common configuration components.

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Next we will need to configure the gateway in

CallManager:

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!Tip - If you have set or been asked to set a domain name on the router the

device name for MGCP would be the router name plus the domain name. So in

our case it could be “HQ_Rtr.voiceie.com” if we had “voiceie.com” as a

domain name on the router.

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2. In order to enable the BR1 router as an H.323 gateway we

need the following configuration on the BR1 router:

BR1(config)#network-clock-participate wic 1

BR1(config)#isdn switch-type primary-ni

BR1(config)#voice-card 0

BR1(config-voicecard)#dsp services dspfarm

BR1(config-voicecard)#exit

BR1(config)#voice translation-rule 1

BR1(cfg-translation-rule)#rule 1 /\(^702333\)\(2...\)/ /\2/

BR1(cfg-translation-rule)#exit

BR1(config)#voice translation-profile STRIP

BR1(cfg-translation-profile)#translate called 1

BR1(cfg-translation-profile)#exit

BR1(config)#voice-port 0/1/0:23

BR1(config-voiceport)#translation-profile incoming STRIP

BR1(config-voiceport)#exit

BR1(config)#controller T1 0/1/0

BR1(config-controller)#framing esf

BR1(config-controller)# linecode b8zs

BR1(config-controller)# pri-group timeslots 1-3,24

BR1(config-controller)# exit

BR1(config)#interface Loopback0

BR1(config-if)#h323-gateway voip interface

BR1(config-if)#h323-gateway voip bind srcaddr 10.10.32.2

BR1(config-if)#exit

BR1(config)#interface Serial0/1/0:23

BR1(config-if)#isdn outgoing display-ie

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Next we will need to setup BR1 as an H.323 gateway in

CallManager:

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!Tip - The most common issue with H.323 gateways is that IP address specified

in CallManager is not the IP address that the H.323 process in the router is

binding. Make sure these IP address match!

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3. In order to enable the BR2 router as an H.323 gateway we

need the following configuration on the BR2 router:

BR2(config)#network-clock-participate wic 0

BR2(config)#isdn switch-type primary-net5

BR2(config)#voice translation-rule 1

BR2(cfg-translation-rule)#rule 1 /\(^44207630\)\(3...\)/ /\2/

BR2(cfg-translation-rule)#exit

BR2(config)#voice translation-profile STRIP

BR2(cfg-translation-profile)#translate called 1

BR2(cfg-translation-profile)#exit

BR2(config)#voice-card 0

BR2(config-voicecard)#dspfarm

BR2(config-voicecard)#dsp services dspfarm

BR2(config-voicecard)#exit

BR2(config)#controller E1 0/0/0

BR2(config-controller)#pri-group timeslots 1-3,16

BR2(config-controller)#exit

BR2(config)#interface Serial0/0/0:15

BR2(config-if)#isdn outgoing display-ie

BR2(config-if)#exit

BR2(config)#voice-port 0/0/0:15

BR2(config-voiceport)#translation-profile incoming STRIP

4. In order to setup the gatekeeper we need to perform the

following configuration on the HQ router:

HQ_Rtr(config)#gatekeeper

HQ_Rtr(config-gk)#zone local CM ccbootcamp.com 10.10.32.1

HQ_Rtr(config-gk)#zone local CME ccbootcamp.com

HQ_Rtr(config-gk)#gw-type-prefix 1#* default-technology

HQ_Rtr(config-gk)#no shutdown

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Next we need to enable CallManager to register to the

gatekeeper through a H.225 gatekeeper controlled

trunk. We can accomplish this by adding the gatekeeper

and then the trunk in CallManager:

!Tip - If you would like to lower the time it takes for gatekeeper to register

after a loss of connectivity then change the “Registration Retry Timeout” to 30

seconds.

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Now we need to add the trunk:

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! Tip - You can validate registration with the “show gatekeeper endpoints”

command on the HQ router.

HQ_Rtr#show gatekeeper endpoints

GATEKEEPER ENDPOINT REGISTRATION

================================

CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags

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

10.1.200.21 33072 10.1.200.21 32794 CM H323-GW

H323-ID: BR2_1

Voice Capacity Max.= Avail.= Current.= 0

10.1.200.25 33077 10.1.200.25 32785 CM H323-GW

H323-ID: BR2_2

Voice Capacity Max.= Avail.= Current.= 0

Total number of active registrations = 4

HQ_Rtr#

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Next we will need to configure the BR2 router is

register with the gatekeeper. This requires the

following configuration:

BR2(config)#interface Loopback0

BR2(config-if)#h323-gateway voip interface

BR2(config-if)#h323-gateway voip id CME ipaddr 10.10.32.1 1719

BR2(config-if)#h323-gateway voip h323-id CME

BR2(config-if)#h323-gateway voip tech-prefix 1#

BR2(config-if)#h323-gateway voip bind srcaddr 10.10.32.3

BR2(config-if)#exit

BR2(config)#gateway

! Tip - Once again you can verify with the “show gatekeeper endpoints”

command on the HQ router.

HQ_Rtr#show gatekeeper endpoints

GATEKEEPER ENDPOINT REGISTRATION

================================

CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags

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

10.10.32.3 1720 10.10.32.3 57205 CM H323-GW

H323-ID: CME

Voice Capacity Max.= Avail.= Current.= 0

10.1.200.21 33072 10.1.200.21 32794 CM H323-GW

H323-ID: BR2_1

Voice Capacity Max.= Avail.= Current.= 0

10.1.200.25 33077 10.1.200.25 32785 CM H323-GW

H323-ID: BR2_2

Voice Capacity Max.= Avail.= Current.= 0

Total number of active registrations = 4

HQ_Rtr#

!Note - You will notice that the directory numbers from the SIP CME did not

register with gatekeeper. This is to be expected as SIP CME phones cannot be

auto registered to gatekeeper.

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Call Routing

1. The has already been accomplished for HQ and BR1 when

we setup the phones previously by giving them the

proper calling search space. To enable class of

service for the BR2 site we need to create the

following COR list:

dial-peer cor custom

name Local

name LD

name INT

name Emergency

dial-peer cor list Emergency

member Emergency

dial-peer cor list Local

member Local

dial-peer cor list Int

member INT

dial-peer cor list LD

member LD

dial-peer cor list Phone1

member Local

member LD

member INT

member Emergency

dial-peer cor list Phone2

member Local

member LD

member Emergency

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Now that we have created the COR list we need to

assign the proper COR list to the voice register pool

of the proper phone:

BR2(config)#ephone-dn 1

BR2(config-ephone-dn)#cor incoming Phone1

BR2(config)#ephone-dn 2

BR2(config-ephone-dn)#cor incoming Phone2

2. In order to accomplish the next two tasks we need to

create all the route groups and route lists first. We

will first need to create the route groups:

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Then next the route lists in the following tables with

the proper settings:

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%&''#=$+&,-.$5,&)"67,8$2&"3$ $

+,#6(1$9(*(-"$:4;-*7()*$%&''"<$ A??QQBA@RSA$

2#8C#,"$ DAED!AE9D?FAGHIJK-,$:L''$+7,-"<$

$$

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!"#$%&'()$&

!"#$%&'()$& *))"+(*$%,&!"#$%&-."#/&0($1&2%23%.)&

!

4567!869:''7;<=>>&

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9:9;!<&*=(! 8:/>6:/!

?+@AB#C! 5!

!!

7!869!?@A&

9@BB'+D!6@&)E!2&@+$F*&G@)'*+$!

:$#!9@BB'+D!6@&)EH$!?I)#&+@B!;@$J! 3+!

9@BB'+D!6@&)E!2&@+$F*&G!;@$J! !

6&#F'I!"'D')$!K3=)D*'+D!9@BB$L! !

9@BB#C!6@&)E!2&@+$F*&G@)'*+$!

"'$%@&C!"'D')$! 6&#"*)!

9@BB#C!6@&)E!2&@+$F*&G!;@$J! !

6&#F'I!"'D')$!K3=)D*'+D!9@BB$L! M!

;#GA#&$! /01!K7BB!6*&)$L!

!

456-B&

9@BB'+D!6@&)E!2&@+$F*&G@)'*+$!

:$#!9@BB'+D!6@&)EH$!?I)#&+@B!;@$J! 3+!

9@BB'+D!6@&)E!2&@+$F*&G!;@$J! !

6&#F'I!"'D')$!K3=)D*'+D!9@BB$L! !

9@BB#C!6@&)E!2&@+$F*&G@)'*+$!

"'$%@&C!"'D')$! 6&#"*)!

9@BB#C!6@&)E!2&@+$F*&G!;@$J! !

6&#F'I!"'D')$!K3=)D*'+D!9@BB$L! NOOPP1NQ!

;#GA#&$! 8NR8:NR"8O>NS,-.0)&!K7BB!6*&)$L!

!!!

456C@9D!@=9C:@='&

"#$%&'()'*+! ,-.TU2?0U72T3U74!

9:9;!<&*=(! 8:/>6:/!

?+@AB#C! 5!

!!

456-B&

9@BB'+D!6@&)E!2&@+$F*&G@)'*+$!

:$#!9@BB'+D!6@&)EH$!?I)#&+@B!;@$J! 3+!

9@BB'+D!6@&)E!2&@+$F*&G!;@$J! !

6&#F'I!"'D')$!K3=)D*'+D!9@BB$L! !

9@BB#C!6@&)E!2&@+$F*&G@)'*+$!

"'$%@&C!"'D')$! 6&#"*)!

9@BB#C!6@&)E!2&@+$F*&G!;@$J! !

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!"#$%&'()$&

!"#$%&'()$& *))"+(*$%,&!"#$%&-."#/&0($1&2%23%.)&

!"#$%&'(%)%*+',-.*)/%0)'1233+4' '

5#67#"+' 89:8;9:(8<=9>?@AB*"',C33'!/"*+4'

'''

456'7&

(#+D"%E*%/0' ?@AF('

1;15'G"/.E' 8;H=!;H'

I0273#J' K'

''

456-8&

1233%0)'!2"*L'M"20+$/"62*%/0+'

;+#'1233%0)'!2"*LN+'I&*#"023'52+O' -0'

1233%0)'!2"*L'M"20+$/"6'52+O' '

!"#$%&'(%)%*+',-.*)/%0)'1233+4' '

1233#J'!2"*L'M"20+$/"62*%/0+'

(%+D2"J'(%)%*+' !"#(/*'

1233#J'!2"*L'M"20+$/"6'52+O' '

!"#$%&'(%)%*+',-.*)/%0)'1233+4' '

5#67#"+' 89:8;9:(8<=9>?@AB*"',C33'!/"*+4'

'

9!:6-8&

1233%0)'!2"*L'M"20+$/"62*%/0+'

;+#'1233%0)'!2"*LN+'I&*#"023'52+O' -0'

1233%0)'!2"*L'M"20+$/"6'52+O' '

!"#$%&'(%)%*+',-.*)/%0)'1233+4' '

1233#J'!2"*L'M"20+$/"62*%/0+'

(%+D2"J'(%)%*+' !"#(/*'

1233#J'!2"*L'M"20+$/"6'52+O' '

!"#$%&'(%)%*+',-.*)/%0)'1233+4' P'

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'''

456';<='&

(#+D"%E*%/0' ?@AF-1CF'

1;15'G"/.E' 8;H=!;H'

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1233%0)'!2"*L'M"20+$/"62*%/0+'

;+#'1233%0)'!2"*LN+'I&*#"023'52+O' -0'

1233%0)'!2"*L'M"20+$/"6'52+O' '

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!"#$%&'()$&

!"#$%&'()$& *))"+(*$%,&!"#$%&-."#/&0($1&2%23%.)&

! "#$%&'!(&)&*+!,-.*)/&0)!1233+4! !

1233$5!"2#*6!7#20+%/#82*&/0+!

(&+92#5!(&)&*+! "#$(/*!

1233$5!"2#*6!7#20+%/#8!:2+;! !

"#$%&'!(&)&*+!,-.*)/&0)!1233+4! !

:$8<$#+! =>?=@>?(=AB>CDEFG*#!,H33!"/#*+4!

!

4!56-7&

1233&0)!"2#*6!7#20+%/#82*&/0+!

@+$!1233&0)!"2#*6I+!J'*$#023!:2+;! -0!

1233&0)!"2#*6!7#20+%/#8!:2+;! !

"#$%&'!(&)&*+!,-.*)/&0)!1233+4! !

1233$5!"2#*6!7#20+%/#82*&/0+!

(&+92#5!(&)&*+! "#$(/*!

1233$5!"2#*6!7#20+%/#8!:2+;! !

"#$%&'!(&)&*+!,-.*)/&0)!1233+4! KAL>M!

:$8<$#+! A>NA>NOMNM!,H33!"/#*+4!

!!!

!

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A summary of all the route lists is below:

We can now create all the route patterns for the HQ

call routing and point them to the proper route list:

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! Note - All the digit manipulation and phone number masking is done in the

route lists and therefore should not be done in the route pattern.

Next we need to add the proper call routing to the

gatekeeper on the HQ gateway:

HQ_Rtr(config-gk)#zone prefix CME 3...

HQ_Rtr(config-gk)#zone prefix CM 7....

HQ_Rtr(config-gk)#zone prefix CME 96......

3. We can now use the previous created route groups and

route lists to setup the BR1 call routing by adding

the following route patterns:

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Now since the BR1 gateway is an H.323 gateway we need

to add the proper dial-peers to the BR1 gateway to

complete the call routing. We may also want to take

into account the SRST task for maintaining class of

service and create the COR list at the same time:

dial-peer cor custom

name Local

name LD

name INT

name Emergency

dial-peer cor list Emergency

member Emergency

dial-peer cor list Local

member Local

dial-peer cor list Int

member INT

dial-peer cor list LD

member LD

dial-peer cor list Phone1

member Local

member LD

member INT

member Emergency

dial-peer cor list Phone2

member Local

member LD

member Emergency

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We can now create the dial-peers with the COR already

configured:

dial-peer voice 1 pots

incoming called-number .

direct-inward-dial

dial-peer voice 2000 voip

preference 1

destination-pattern 2...

session target ipv4:10.1.200.25

ip qos dscp cs3 signaling

dial-peer voice 2001 voip

preference 2

destination-pattern 2...

session target ipv4:10.1.200.21

ip qos dscp cs3 signaling

dial-peer voice 7 pots

corlist outgoing Local

destination-pattern 9[2-9]......

port 0/1/0:23

forward-digits 7

dial-peer voice 11 pots

corlist outgoing LD

destination-pattern 91[2-9]..[2-9]......

port 0/1/0:23

forward-digits 11

!

dial-peer voice 911 pots

corlist outgoing Emergency

destination-pattern 911

port 0/1/0:23

forward-digits all

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!

dial-peer voice 9911 pots

corlist outgoing Emergency

destination-pattern 9911

port 0/1/0:23

forward-digits 3

!

dial-peer voice 9011 pots

corlist outgoing Int

destination-pattern 9011T

port 0/1/0:23

prefix 011

4. We can now setup the call routing for BR2 with the

following configuration on the BR2 router. We first

need to allow all the communications that maybe needed

in our environment:

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

supplementary-service h450.12

no supplementary-service sip refer

fax protocol cisco

h323

sip

registrar server expires max 1200 min 300

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Now we can create all the needed dial-peers with the

appropriate COR list which were previously defined:

dial-peer voice 1 pots

incoming called-number .

direct-inward-dial

!

dial-peer voice 2 pots

corlist outgoing Local

destination-pattern 90[1-9].......

port 0/0/0:15

prefix 0

!

dial-peer voice 1000 voip

preference 1

destination-pattern 71...

session target ras

!

dial-peer voice 1001 pots

preference 2

destination-pattern 71...

port 0/0/0:15

prefix 0017029461

!

dial-peer voice 2000 voip

preference 1

destination-pattern 72...

session target ras

!

dial-peer voice 2001 pots

preference 2

destination-pattern 72...

port 0/0/0:15

prefix 0017023332

!

dial-peer voice 7 pots

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corlist outgoing Local

destination-pattern 9[1-9]......

port 0/0/0:15

forward-digits 7

!

dial-peer voice 999 pots

corlist outgoing Emergency

destination-pattern 999

port 0/0/0:15

forward-digits all

!

dial-peer voice 900 pots

corlist outgoing Int

destination-pattern 900.T

port 0/0/0:15

prefix 00

We also need to add a translation pattern to strip the

“7” as the call comes into callmanager from BR2:

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We will also need to enable transcoding on the BR2

gateway to support calls transgressing from HQ/BR1 to

BR2:

sccp local FastEthernet0/0.230

sccp ccm 10.1.230.1 identifier 1 priority 1 version 3.1

sccp

sccp ccm group 1

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associate ccm 1 priority 1

associate profile 1 register TRANSCODE

dspfarm profile 1 transcode

codec g711ulaw

codec g711alaw

codec g729r8

codec g729abr8

codec g729ar8

codec g729br8

maximum sessions 4

associate application SCCP

telephony-service

sdspfarm units 2

sdspfarm transcode sessions 4

sdspfarm tag 1 TRANSCODE

!Tip - At this point you can start testing your calls to make sure they all

complete correctly. In order to test redundancy you can shutdown the

gatekeeper or the corresponding T1/E1 controller to test.

Time Saver - By saving the testing until your call routing is complete you

can save time by not interrupting your configuration and then fixing as you

test.

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Codec and Call Admission Control

1. This task was already accomplished earlier when we

created the regions.

2. This task was already accomplished earlier when we

created the regions.

3. This task was completed when we create the locations

earlier.

4. In order to accomplish this task we need to enable

zone bandwidth control on the gatekeeper:

HQ_Rtr(config)#gatekeeper

HQ_Rtr(config-gk)#bandwidth total zone CME 48

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High Availability Features

1. In order to accomplish this task we need to enable

“call-manager-fallback” in the BR1 gateway:

application

global

service alternate DEFAULT

call-manager-fallback

secondary-dialtone 9

max-conferences 4 gain -6

ip source-address 10.1.130.1 port 2000

max-ephones 2

max-dn 10

transfer-pattern .T

voicemail 917029461170

call-forward pattern .T

2. This task was accomplished earlier by setting the BR1

phone 1 device pool to “BR1-NOSRST” which had the SRST

reference disabled.

3. This was completed during the call routing section as

the dial-peers were created with 9 at the beginning.

4. In order to accomplish this we need to add the proper

COR list to the DN of BR1 phone 2 in the incoming

direction:

call-manager-fallback

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cor incoming Phone2 1 2002

5. Most of this task has already been completed during

the phone setup section. We created the AAR groups

with the appropriate prefixes and then assigned the

DN’s to the appropriate AAR group. The last step is to

enable the AAR service via the callmanager service

parameters:

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Media Resources Management

1. In order to accomplish this task we first need to

create the resource in callmanager:

Next we need to configure the SCCP and DSPFARM

settings on the HQ router:

sccp local Loopback0

sccp ccm 10.1.200.21 identifier 2 priority 2 version 7.0

sccp ccm 10.1.200.25 identifier 1 priority 1 version 7.0

sccp

!

sccp ccm group 1

associate ccm 1 priority 1

associate ccm 2 priority 2

associate profile 2 register HQTRANS

associate profile 1 register HQCONF

!

dspfarm profile 2 transcode

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codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

maximum sessions 4

associate application SCCP

!

dspfarm profile 1 conference

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

codec g729r8

codec g729br8

maximum sessions 4

associate application SCCP

!Tip - You can use the “show sccp” and the show “dspfarm profile” command

to verify gateway configuration. You may need to disable and then enable the

SCCP function if the resources aren’t registering.

2. In order to accomplish this task we first need to

create the resource in callmanager:

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Time Saver - This is another great instance of where copy and paste can

help you as the configuration will be very similar to the HQ router.

sccp local Loopback0

sccp ccm 10.1.200.25 identifier 1 priority 1 version 7.0

sccp ccm 10.1.200.21 identifier 2 priority 2 version 7.0

sccp

!

sccp ccm group 1

associate ccm 1 priority 1

associate ccm 2 priority 2

associate profile 1 register BR1CONF

associate profile 2 register BR1TRANS

!

dspfarm profile 2 transcode

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

codec gsmfr

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maximum sessions 4

associate application SCCP

!

dspfarm profile 1 conference

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

codec g729r8

codec g729br8

maximum sessions 4

associate application SCCP

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3. To accomplish this task we first need to create the

resource in callmanager:

The gateway configuration portion was accomplished

above in the conference bridge section.

4. To accomplish this task we first need to create the

resource in callmanager:

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The gateway configuration portion was accomplished

above in the conference bridge section.

5. In a previous step we create all the media resource

groups and list. Now we can add the proper media

resources to the proper groups. The publisher MOH

server will need to be put into the “PUB_MOH” media

resource group and we will need to make sure that this

is the MOH only server listed in the media resource

list for HQ.

6. We need to perform the same actions for BR1 but with

the subscriber server being put into the “SUB_MOH”

media resource group and making sure that this is the

only MOH server listed in the media resource list for

BR1. In addition we will need to enable the default

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audio source, subscriber MOH server, and the “SUB_MOH”

media resource group for multicasting:

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We need to insure that the “Max Hops” is set to “1” so

that the multicast stream stays local.

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Now we need to enable the G.729 codec through the

Cisco IP Voice Media Streaming App service parameters:

We then need to configure the BR1 gateway to support

multicast and the router to play the MOH file from

flash:

call-manager-fallback

moh music-on-hold.au

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multicast moh 239.1.1.1 port 16388 route 10.1.130.1

ip multicast-routing

interface Vlan130

ip pim sparse-dense-mode

! Tip - To test place a call from HQ to BR1 and then put the call on hold from

both ends. You should receive MOH but hear different music on both sides.

7. In order to accomplish this we first need to create

the MeetMe conference resources in callmanager:

We then need to change the callmanager service

parameter to allow for 10 participants per conference:

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8. This is accomplished through the same callmanager

service parameter section as the task above by setting

the “Drop Ad Hoc Conference” parameter correctly.

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QoS Features

1. We can accomplish this with the following

configuration.

On the HQ gateway:

interface Serial0/2/0.1 point-to-point

bandwidth 384

frame-relay interface-dlci 201

auto qos voip trust fr-atm

On the BR1 Gateway:

interface Serial0/2/0.1 point-to-point

bandwidth 384

frame-relay interface-dlci 101 ppp

auto qos voip trust fr-atm

The on both gateways modify the auto generated policy

map per the requirements:

policy-map AutoQoS-Policy-Trust

class AutoQoS-VoIP-RTP-Trust

priority percent 33

class AutoQoS-VoIP-Control-Trust

bandwidth percent 5

class class-default

fair-queue

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2. We can accomplish this with the following

configuration.

On the HQ gateway:

interface Serial0/2/0.2 point-to-point

bandwidth 1200

frame-relay interface-dlci 202

auto qos voip trust

On the BR2 gateway:

interface Serial0/2/0.1 point-to-point

bandwidth 1200

frame-relay interface-dlci 102

auto qos voip trust

Then on both gateways modify the auto generated policy

map per the requirements:

policy-map AutoQoS-Policy-Trust2

class AutoQoS-VoIP-RTP-Trust

priority 360

class AutoQoS-VoIP-Control-Trust

bandwidth 60

class class-default

fair-queue

3. This was accomplished earlier under the two dial-peers

pointing traffic back to the callmanagers.

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Application and End Users

In order to move forward past this section we need to

define all the users. Outlined below in tables are the

users that need to be created and the settings for those

users.

!Tip - Pay close attention to the associated devices and rights of each user.

Before you start to add user you will need to create a “Standard AXL User”

group and assign the “Standard AXL API Access” role to that group.

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!""#$%&'$()*+,-.*

+,-.*/0* /)1(.2&'$()*

!"#$%&&

'(()#*+"#,-&./01&2-3,14+"#,-&

510/0-*0&$1,6(& 7"+-8+18&510/0-*0&91,6(&

'**0("&510/0-*0&76:/*1#("#,-& ;&

'**0("&<6"=,3=8#+),9&>?@?>& ;&

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!'5@&2-3,14+"#,-&

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>,)0/& 7"+-8+18&!M2&')),%&!,-"1,)&,3&'))&B0C#*0/&7"+-8+18&!M2&?-+:)08&

&&

25!!&&

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510/0-*0&$1,6(& 7"+-8+18&510/0-*0&91,6(&

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'**0("&<6"=,3=8#+),9&>?@?>& ;&

'**0("&.-/,)#*#"08&;,"#3#*+"#,-& ;&

'**0("&>0()+*0/&A0+801& ;&

B0C#*0&2-3,14+"#,-&

!,-"1,))08&B0C#*0/& &

!'5@&2-3,14+"#,-&

'//,*#+"08&!'5@&51,3#)0/& &

5014#//#,-/&2-3,14+"#,-&

$1,6(/& 7"+-8+18&'NO&./01&

>,)0/& 7"+-8+18&'NO&'52&'**0//&

&&

.-#"P&&

'(()#*+"#,-&./01&2-3,14+"#,-&

510/0-*0&$1,6(& 7"+-8+18&510/0-*0&91,6(&

'**0("&510/0-*0&76:/*1#("#,-& ;&

'**0("&<6"=,3=8#+),9&>?@?>& ;&

'**0("&.-/,)#*#"08&;,"#3#*+"#,-& ;&

'**0("&>0()+*0/&A0+801& ;&

B0C#*0&2-3,14+"#,-&

!,-"1,))08&B0C#*0/& &

!'5@&2-3,14+"#,-&

'//,*#+"08&!'5@&51,3#)0/& &

5014#//#,-/&2-3,14+"#,-&

$1,6(/& 7"+-8+18&'NO&./01&

>,)0/& 7"+-8+18&'NO&'52&'**0//&

&

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The next list of Application users is created during the

setup of the contact center server and therefore will not

need to be create:

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!""#$%&'$()*+,-.*

+,-.*/0* /)1(.2&'$()*

!

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)%%*&+$#&,-!./01!2-3,14$#&,-!

510/0-+0!61,7%! 8#$-9$19!510/0-+0!:1,7%!

)++0%#!510/0-+0!87;/+1&%#&,-! <!

)++0%#!=7#>,3>9&$*,:!?@A@?! <!

)++0%#!.-/,*&+&#09!<,#&3&+$#&,-! <!

)++0%#!?0%*$+0/!B0$901! <!

C0D&+0!2-3,14$#&,-!

E,-#1,**09!C0D&+0/! ))('F'G!))('F'H!))('F'I!))('F'J!))('F'K!))!2EC!

E)5A!2-3,14$#&,-!

)//,+&$#09!E)5A!51,3&*0/! !

5014&//&,-/!2-3,14$#&,-!

61,7%/! 8#$-9$19!EL2!@-$;*09!

?,*0/! 8#$-9$19!EL2!@-$;*09!

!!

14"#$%&'!!

)%%*&+$#&,-!./01!2-3,14$#&,-!

510/0-+0!61,7%! 8#$-9$19!510/0-+0!:1,7%!

)++0%#!510/0-+0!87;/+1&%#&,-! <!

)++0%#!=7#>,3>9&$*,:!?@A@?! <!

)++0%#!.-/,*&+&#09!<,#&3&+$#&,-! <!

)++0%#!?0%*$+0/!B0$901! <!

C0D&+0!2-3,14$#&,-!

E,-#1,**09!C0D&+0/! 8@5FF'HEIMCNCEE!8@5FFNFKOEMIJHO!8@5FFNFKOEMC)NJ!8@5FFNFKOENCEAK!

E)5A!2-3,14$#&,-!

)//,+&$#09!E)5A!51,3&*0/! !

5014&//&,-/!2-3,14$#&,-!

61,7%/! 8#$-9$19!EL2!@-$;*09!

?,*0/! 8#$-9$19!EL2!@-$;*09!

!!

!

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The following End Users will also need to be created with

the correct settings from the table:

!"#$%&'($)*+"#'"&'#,$

%&'($-.$ /'0'12+"'$

3456'($

*+"7(+00'#$

.'89:'&$

*+"7(+00'#$

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=&&+:9>7'#$

!?7'"&9+"&$

@(+41&$ A+0'&$

!"#$ %&&#$ '()&&*&+,-*.-/+$ $$ %&&#$01$)23!"#341567189$%&&#$3$4)--$

'581:87:$--;$(1:$<=67=$'581:87:$-24$(18>96:$

'581:87:$--;$(1:$<=67=$'581:87:$--;<'("$?:@010=57850A1$'581:87:$-24$(18>96:$

!"%$ %&&%$ '()&&#B-C%.*.--$ $$ %&&%$01$)23!"#341567189$

'581:87:$--;$(1:$<=67=$'581:87:$-24$(18>96:$

'581:87:$--;$(1:$<=67=$'581:87:$--;<'("$?:@010=57850A1$'581:87:$-24$(18>96:$

DE#$ #&&#$ '()&&*&+,-%CFB,$<)-DE#$

$$ #&&#$01$)23DE341567189$#&&#$3$4)--$

'581:87:$--;$(1:$<=67=$'581:87:$-24$(18>96:$

'581:87:$--;$(1:$<=67=$'581:87:$--;<'("$?:@010=57850A1$'581:87:$-24$(18>96:$

DE%$ #&&%$ '()&&*&+,-%.?*F$ $$ #&&%$01$)23DE341567189$

'581:87:$--;$(1:$<=67=$'581:87:$-24$(18>96:$

'581:87:$--;$(1:$<=67=$'581:87:$--;<'("$?:@010=57850A1$'581:87:$-24$(18>96:$

G7=8:@01$ $$ $$ $$ H$IA16$J$ $$ $$

KAL1$ #&&C$ $$ MAL1$ #&&C$01$)23DE341567189$

'581:87:$--;$(1:$<=67=$'581:87:$-24$(18>96:$

'581:87:$--;$(1:$<=67=$'581:87:$--;<'("$?:@010=57850A1$'581:87:$-24$(18>96:$

!

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Voicemail Integrations

1. In order to accomplish this we first need to perform

all the required setup in callmanager for voicemail.

We start first with creating the voicemail ports:

Time Saver - You can create the ports manually or through the wizard.

Obviously the wizard is designed to save time as it creates the line group

for you will all the voicemail ports as members.

Below is the line group configuration:

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Next you will need to add this line group to a hunt list:

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Now we need to create the voicemail pilot (Hunt Pilot)

and point it to the correct hunt list:

We can now create the MWI on and MWI off extensions:

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We can now set the default voicemail pilot and

profile:

!Tip - You will see that we create a profile named strip. This will be used later

in task 5 in order to strip the “*” out of the extension before it rolls to

voicemail. The settings of this profile are shown below.

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Now in order to start the integration with Unity

Connections we will first need to create a port group:

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Next will need to define the callmanager servers Unity

will use through the Edit->Servers page under the port

group:

During the port group creation we were asked how many

ports and the device name prefix. This created the

ports for us automatically:

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Now we can finish the integration by settings the

parameters for the phone system to include the AXL

servers for importing users from callmanager:

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! Tip - This is also where the loop detection is enabled which is required in the

next task.

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Now we can define the AXL servers through the Edit->

Cisco Unified Communications Manager AXl Servers under

the phone system:

!Note - You will see the application user “Unity” which we create previously.

This user must be in the “Standard AXL User Group” that we created in order

to import users from callmanager.

2. In Unity Connections loop prevention is on by default.

However, you should be aware that there are two ways to

accomplish loop prevention. One is by extension and the

other is through DTMF tone based recognition.

3. In order to accomplish this task you must first modify

the default voicemail user template to change the default

password. You can then import all the users into Unity:

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You can now go to the import users page an import all

users:

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When finished your users page should look like the

following:

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! Tip - If the primary extension of the user is not set then the user will not show

up for import.

4. In order to configure the auto attendant we will first

need to create the appropriate CTI route point in

callmanager and forward all calls to voicemail for this

CTI route point:

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Time Saver -You can see two CTI route points listed as the second one will

be used to record the greeting for the call handler in this task.

We can setup the Greetings Administrator access for HQ

phone 1 user through the call routing section:

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Once this is complete we can now setup the call

handler for the auto attendant:

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Next we can setup the caller input options as required

in the task:

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We now need to enable and setup the closed extension

to take a message:

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We also need to modify the standard greeting so that a

message is not taken:

We need to also assign the call handler owner so that

the HQ phone 1 user can access the call handler and

change the greeting via the Greetings Administrator:

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5. Part of this task has already been configured during the

phone setup section and callmanager voicemail section.

What we need to double check is that the voicemail

profile named “STRIP” is assigned to the BR1 phone 2

extension of “*2002”:

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6. In order to accomplish this task we first need to define

access to the CUE module, the ephone-dn’s for MWI, and

then dial-peers for the correct number(s) that need to be

forwarded to CUE:

interface Service-Engine1/0

ip unnumbered FastEthernet0/0.230

service-module ip address 10.1.230.2 255.255.255.0

service-module ip default-gateway 10.1.230.1

ip route 10.1.230.2 255.255.255.255 Service-Engine1/0

ephone-dn 4

number 8001....

mwi on

ephone-dn 5

number 8002....

mwi off

dial-peer voice 3180 voip

destination-pattern 3180

session protocol sipv2

session target ipv4:10.1.230.2

dtmf-relay rtp-nte

codec g711ulaw

no vad

!

dial-peer voice 3000 voip

destination-pattern 3000

session protocol sipv2

session target ipv4:10.1.230.2

dtmf-relay rtp-nte

codec g711ulaw

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no vad

Time Saver - In order to save time we can go ahed and create the dial-peer

need for task 8.

You can now start the CUE initialization process by

session into the CUE module from the BR2 gateway with

the following command:

BR2#service-module service-engine 1/0 session

Once you have completed the setup the CUE module will

reboot and is ready for configuration via the web

interface or via command line. You will now need to

set the MWI on and MWI off:

ccn application ciscomwiapplication aa description "ciscomwiapplication" enabled maxsessions 8 script "setmwi.aef" parameter "CallControlGroupID" "0" parameter "strMWI_OFF_DN" "8002" parameter "strMWI_ON_DN" "8001" end application

You can test access by pressing the voicemail button

on any BR2 phone.

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7. You can accomplish this via the web interface or via the

command line. We will show through the command line:

username br2phone3 create username BR2Phone2 create username BR2Phone1 create username admin create

username BR2Phone2 phonenumber "3002"

username BR2Phone1 phonenumber "3001"

username BR2Phone2 phonenumberE164 "442076303002"

username BR2Phone1 phonenumberE164 "442076303001"

voicemail mailbox owner "BR2Phone1" size 21176 description "BR2Phone1 mailbox" end mailbox voicemail mailbox owner "BR2Phone2" size 21176 description "BR2Phone2 mailbox" end mailbox

! Tip - You can test access and MWI by leaving a message and making sure the

MWI is working for each mailbox.

8. Part of this task was completed in task 6 and now we just

need to define the auto attendant in CUE:

ccn trigger sip phonenumber 3000

application "autoattendant"

enabled

maxsessions 8

end trigger

9. The was already accomplished earlier during the call

routing section. You will just need to test!

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IPCC Express Voice Application

1. In order to accomplish this task you will need to

complete the normal IPCC Express setup. Once this is

done you will need to create a Call Control Group

based on the task requirements:

!Tip - If required you can click the “Show More...” button and set the partition,

calling search space, AAR group, and other settings that may be required. In

this case we would probably want to set things like the device pool and

location.

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2. To accomplish this task you will need to create a new

application and trigger using the default auto

attendant script:

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3. First we need to create the application and trigger:

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We now can modify the resources and assign the skills

required for task 5:

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! Tip - If you don’t see any resources in IPCC Express then you probably

haven’t assigned the users an IPCC extension in callmanager.

We now need to create the contact services queue with

the same name we referenced under the application:

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We now need to create the IP Agent Phone Service:

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We can then assign the IP phone service to the proper

phones:

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You should now be able to login to the IP Agent Phone

service and test calls into the queue.

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4. In order to accomplish this we can modify the default

script to place the call on hold and then off hold

during the queuing process:

5. This was accomplished in task 3 by assigning the users

skills and then modifying the queue to route based on

“Most Skilled”.

6. This is shown in the script example provided under

task 4.

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! Tip - You should get to know the “Get Reporting Statistics” function in IPCC

Express. The function will allow you to create checks for just about any task.

CallManager Voice Application

1. Part of this task was already completed when we created

the user in callmanager. We now need to create the IP

phone service:

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Next we can create the device profile:

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We need to make sure we subscribe the device profile

to the extension mobility phone service:

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We then need to enable the required phones for extension

mobility and subscribe them to the IP phone service for

extension mobility:

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Presence

1. The first step is to run through the setup for

Presence after the initial install. Once this is

complete we can setup the required SIP trunk in

callmanager and set the SIP publish trunk in the

callmanager service parameters:

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Now set the CUP Publish Trunk in the callmanager

service parameters:

We can now create the CUPC in callmanager:

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Make sure on all the phones and CUPC in callmanager

that the subscribe calling search space is set

accordingly:

Next we need to assign the capabilities in

callmanager:

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Once this is complete we now need to finish the

configuration of the presence server. We can start by

validating the setup via System->CUCM Publisher:

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You can now configure the incoming and outgoing ACL’s:

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We now need to configure the CM SIP Publish trunk

under the Presence->Settings:

We now need to configure the callmanager as a presence

gateway:

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We now need to modify the proxy services settings:

We can then start configuring the CUPC settings:

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Next we need to configure the Unity server as a

voicemail server and then configure the voicemail

profile:

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We can now set the proper profiles for the HQ phone 1

user:

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Next we need to configure the “Deskphone Control

Settings” so that the CUPC client can control the

phone:

! Tip - The phones that need to be controlled by the “CtiGw” user were defined

earlier during the Application User section.

! Tip - You will also want to make sure that the phone being controlled has an

owner assigned and is associated to the end user.

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We can then enable the HQ phone 1 user for desktop

phone control:

Next you will need to the proxy domain and restart the

presence and proxy services:

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!Tip - You will need to create a host file entry on the machine running CUPC

based on the domain name you select in the presence server parameters.

Example: “presence.ccbootcamp.com” pointing to 10.1.200.23.

2. We need to first enable Unified Client access within

Unity Connections:

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Next we need to setup the Unity account user name and

password in the CUPC client and check that all

services are up and running:

!Note - You can switch between modes and check services for each setup. Keep

in mind that the service for the mode your not in will show as down or not

configured.

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Chapter 4 - Lab 4

Brief Lab Overview:

This lab is designed to test your IP-to-IP gateway

knowledge.

Basic Campus Design:

1. Configure VLANS and IP networks in all three sites

according to the following two tables:

!

VLAN HQ BR1 BR2

Servers 1 N/A N/A

Data 20 120 220

Voice 30 130 230

!

Network HQ BR1 BR2

Servers 10.1.200.0/24 N/A N/A

Data 10.YY.20.0/24 10.YY.120.0/24 10.YY.220.0/24

Voice 10.YY.30.0/24 10.YY.130.0/24 10.YY.230.0/24

2. Make sure you set all voice traffic to use the voice

VLAN for all sites.

3. Configure phones on HQ site to have IP addresses

120-130. Use IOS DHCP on the HQ router to accomplish

this task.

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4. Configure BR1 phones to have IP addresses of 130–140

using the IOS DHCP on the BR1 Router.

5. Configure BR2 phones to have IP addresses 210–220

using the IOS DHCP on the BR2 router.

6. Configure HQ router as an NTP master clock with

stratum 2 for local time zone in Los Angeles and

configure BR1 and BR2 to synchronize their clocks to

HQ. BR1 is in New York time zone and BR2 is in India.

CallManager and CallManager Express:

1. Register all phones according to the dial plan table

using SCCP loads. Make sure the phones will display

calling names in every scenario presented (internal

and PSTN):

2. Configure directory number 3003 as a second line on

both phones in BR2. When a call comes into 3003 it

should ring on both phones and be answered by the

first user to respond.

3. Make sure you that when the initiator of a conference

call within the BR2 site drops that the conference

call drops.

4. Make sure all phones display the same general

information aside from the line specific

configuration; this refers to HQ and BR1.

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5. Allow BR2 phone 1 to pickup calls ringing on BR2

phone 2 and vice versa. This should work for internal

and external calls.

6. Both phone 1 and phone 2 are members of VIP sales

group and are required to take calls dialed to

442076303111. The calls are expected to ring the

phones in sequential order. Both phones should ring

for 5 seconds and if the call is not answered it

should be forwarded to DN 3000.

7. Set BR2 phone 1 for extension mobility with an auto-

logout enabled. User BR2 Phone 1 should be able to

login to BR2 phone 1 and have all their settings

enabled.

8. Enable both phones at site BR2 for VoiceView Express.

9. Configure BR2 phone 2 so that when the phone is

called it rings our to 6305000 after 5 seconds and

then forwards to voicemail after 10 seconds.

Voice Gateways and Signaling:

1. Configure the HQ router as a IOS MGCP gateway using T1

PRI and NI as the line parameters.

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2. Configure BR1 router as a IOS H.323 gateway using T1

PRI and NI as the line parameters.

3. Configure BR2 as E1 PRI and make sure all BR2 numbers

are reachable.

4. Configure HQ router as a gatekeeper with the following

details:

Local zone name: CM

Domain name: ccbootcamp.com

Register CallManager and CME to the gatekeeper.

Register the CallManager with a tech-prefix of “1#”

and a zone called “CM”. Register CME with a tech-

prefix of “1#” and to the same zone.

Call Routing:

1. In every site, phone 1 should be allowed to dial

international and below numbers, phone 2 should only

be allowed to call long distance and below.

2. Configure the following dialing options for users in

HQ:

7+4 digits Calls to BR2

911 and 9911 Emergency

9+7 digits

The first is in the range of

2 to 9

Local

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9+1+10 digits

The first and the forth are

in the range of 2 to 9.

Long Distance

9+011+any number of digits

Please allow users to dial

the hash sign at the end of

the number in order to have

faster response

International

• Local calls from HQ should use local HQ gateway

and BR1 gateway as backup.

• Long distance calls from HQ should use local HQ

gateway as first choice and then BR1 gateway.

• Calls to BR1 PSTN numbers should be routed out

BR1 gateway (Toll Bypass), with fallback to local

HQ gateway.

• International calls to BR2 numbers, using the 7+4

digits pattern, should be routed via the

gatekeeper as VoIP and use the local HQ gateway

as backup. These calls should be sent to the

gatekeeper as G711ulaw and received as G729 at

BR2.

• All other International calls should route out

the HQ gateway.

3. Configure the following dialing options for users in

BR1:

7+4 digits Calls to BR2

911 and 9911 Emergency

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9+7 digits

The first is in the range of 2

to 9

Local

9+1+10 digits

The first and the forth are in

the range of 2 to 9.

Long Distance

9+011+any number of digits

Please allow users to dial the

hash sign at the end of the

number in order to have faster

response

International

• Local calls from BR1 should use the local BR1 IOS

gateway and then the HQ gateway as a backup.

• Long distance calls from BR1 should use the BR1

gateway as a first choice and then the HQ gateway

as a backup.

• Calls to HQ PSTN numbers (Area Code+Prefix)

should be routed out the HQ gateway (Toll Bypass)

with fallback to the local BR1 gateway.

• International calls to BR2 numbers, using the 7+4

digits pattern, should be routed via the

gatekeeper as VoIP and use the BR1 gateway as

backup. These calls should be sent to the

gatekeeper as G711 and received as G729 at BR2.

• All other International calls should route out

the BR1 gateway.

4. Configure the following dialing options for users in

BR2:

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7+4 digits Calls to HQ and BR1

999 Emergency

9+7 digits Local

9+0+8 digits Long Distance

9+00+any number of digits

Please allow users to dial

the hash sign at the end of

the number in order to have

faster response

International

• International calls to HQ and BR1 should use the

gatekeeper first and the local PRI as a backup.

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Codec and Call Admission Control:

1. All calls within a site should use the G.711 codec.

2. All calls between sites should use the G.729 codec.

3. Allow two concurrent calls to and from BR1.

4. Make calls from HQ to BR2 arrive at the gatekeeper as

G.711 but traverse the WAN as G.729.

High Availability Features:

1. Configure BR1 as an SRST router.

2. Make sure you only allow BR1 phone 2 to register via

SRST during a failure.

3. Preserve all PSTN dialing with using 9 as the first

digit in the string.

4. When in SRST mode, the BR1 phone 2 is not allowed to

dial international calls.

5. Allow HQ and BR1 phones to call each other in times

when more then 2 concurrent calls are in process

between the sites.

Media Resources Management:

1. Configure conferencing for the HQ site to use hardware

based resources in the HQ gateway first then hardware

resources in the BR1 gateway.

2. Configure conferencing for the BR1 site to use

hardware based resources in the BR1 gateway first then

hardware resources in the HQ gateway.

3. Configure transcoding for the HQ site to use hardware

based resources in the HQ gateway first then hardware

resources in the BR1 gateway.

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4. Configure transcoding for the BR1 site to use hardware

based resources in the BR1 gateway first then hardware

resources in the HQ gateway.

5. Configure MOH for the HQ site based on unicast and

streamed from the publisher CallManager.

6. Configure multicast MOH for BR1, based on G729 codec

from the subscriber, and streamed from the BR1

gateway.

7. Allow 5 simultaneous meet me conferences for the HQ

users, in the number range 1201-1205. Every meet me

conference should allow a maximum of ten participants.

8. Make sure that when the initiator of an Ad-Hoc

conference hangs up, the conference is allowed to

continue as long as an internal user is still

connected.

QoS Features:

10.Configure the frame relay link between HQ and BR1 as

if it is 384k in bandwidth, make sure you enable MLP

LFI on it.

• VoIP signaling should have 5% of the line as

guarantied bandwidth.

• VoIP barrier should have 33% of the line as priority

bandwidth.

• All other traffic should be weighted fair queued.

11.Configure the frame relay link between HQ and BR2 as

if it is 1200k in bandwidth.

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• VoIP signaling should have 60k of the line reserved

bandwidth.

• VoIP barrier should have 360k of the line priority

bandwidth.

• All other traffic should be weighted fair queued.

12.The BR1 gateway should send the signaling and RTP

traffic marked as CS3 and EF.

Voicemail Integrations:

1. Configure voicemail for HQ and BR1 based on the Unity

Connections server using the following parameters:

• Pilot 1170, port 1171-1174, MWI on 1180, MWI off

1181.

2. Make sure you prevent Unity Connections from calling

to itself and creating call loops which consume the

available ports.

3. Configure a voicemail box for phones 1 and 2 in HQ and

BR1 with a phone password of 147369 for all users.

4. Configure an auto attendant for HQ with DN 1000. It

should have a customized greeting that allows dialing

an extension and walking through the users directory

by pressing 1. When users press 0 they should be

forwarded to extension 1001. During closed times,

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assume normal weekday schedule, it should take a

message and store it in the HQ phone 1 mailbox.

5. Create a second line on BR1 phone 2 with extension

*2002. A call coming to the primary line should roll

to this line if the primary is busy. The user should

be able to use this line for voicemail access. Do not

use any alternate extensions.

6. Configure BR2 CME to use the local Unity Express as a

voicemail, using the following parameters:

• Pilot 3180, MWI on 8001, MWI off 8002.

7. Create voicemail for BR2 phone 1 and 2. Use a blank

phone password for all users.

8. Create an auto attendant for BR2 that will work when

DN 3000 is dialed.

9. Make sure that calls from HQ can route via gatekeeper

and hit the auto attendant in BR2.

IPCC Express Voice Application:

1. Configure the IPCC on the Contact Center server to

work with the CallManager using the following

parameters (all passwords are “cisco” and all pin’s

are “12345”):

Jtapi username: jtapi

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Resource manager user: rmjtapi

CTI ports: 1015 – 1019.

2. Configure the auto attendant script to answer when a

call arrives at 1020.

3. Configure an ICD queue that will be triggered when

dialing extension 1021. HQ phone 1 should be used as

an IP phone agent for user “HQ1” with the password

“cisco”. BR1 phone 1 should be used as a IP phone

agent with user “BR1” with a password “cisco”.

4. Callers should hear music on hold while delayed in

queue.

5. Calls should be routed via the queue based on a skill

named “Sales”. HQ phone 1 should be skilled higher

than BR1 phone 1.

6. Before a call is routed to the queue a check should be

performed to see if any agents are logged into the

queue. If no agents are logged in then the call should

be routed to HQ phone 2.

CallManager Voice Application:

1. Create extension mobility for user “john” with a

password of “cisco”. Which will allow him to use

extension 1007 on HQ and BR1 phone 2 in each site.

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Make sure that the extension mobility service is only

available on phone 2 in HQ and BR1.

Presence:

1. Integrate the Presence server with CallManager. Make

sure the HQ Phone 1 user can login via CUPC and

control the desktop phone. HQ Phone 1 user should also

be allowed to switch to softphone mode.

2. Make sure HQ Phone 1 can see the voicemail in their

voicemail box through the CUPC client.

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Solutions:

Basic Campus Design

1. Start with “show vlan” and make sure you have all

VLANS defined in all devices. If not create them and

then add the IP interfaces to each device with the

correct IP subnet from the table provided:

Time Saver - In order to save time you should go ahead and configure the

layer 2 QoS in the HQ switch.

HQ:

HQ-3750#

HQ-3750#config t

Enter configuration commands, one per line. End with CNTL/Z.

HQ-3750(config)#vlan 20!! !

HQ-3750(config-vlan)#name Data

HQ-3750(config-vlan)#exit

HQ-3750(config)#vlan 30!! !

HQ-3750(config-vlan)#name Voice

HQ-3750(config-vlan)#exit

HQ-3750(config)#

HQ-3750(config)#interface range FastEthernet 1/0/6 - 7

HQ-3750(config-if-range)#switchport voice vlan 30

HQ-3750(config-if-range)#switchport access vlan 20

HQ-3750(config-if-range)#switchport trunk encapsulation dot1q

HQ-3750(config-if-range)#switchport mode trunk

HQ-3750(config-if-range)#auto qos voip cisco-phone

HQ-3750(config)#interface range FastEthernet 1/0/1 - 5

HQ-3750(config-if-range)#switchport host

HQ-3750(config-if-range)#auto qos voip trust

HQ-3750(config)#interface FastEthernet 1/0/24

HQ-3750(config-if)#auto qos voip trust

HQ-3750(config)#interface FastEthernet0/0.20

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HQ-3750(config-if)#encapsulation dot1Q 20

HQ-3750(config-if)#ip address 10.1.20.1 255.255.255.0

HQ-3750(config)#exit

HQ-3750(config)#interface FastEthernet0/0.30

HQ-3750(config-if)#encapsulation dot1Q 30

HQ-3750(config-if)#ip address 10.1.30.1 255.255.255.0

! Tip - Don’t forget to add the new networks into the routing protocol.

HQ-3750(config)#router ospf 1

HQ-3750(config-router)#network 10.1.20.0 0.0.0.255 area 0

HQ-3750(config-router)#network 10.1.30.0 0.0.0.255 area 0

BR1:

BR1#vlan database

BR1(vlan)#vlan 120 name Data state active

VLAN 120 added:

Name: Data

State ACTIVE

BR1(vlan)#vlan 130 name Voice state active

VLAN 130 added:

Name: Voice

State ACTIVE

BR1(vlan)#exit

APPLY completed.

Exiting....

! Tip - The “vlan database” command has been deprecated so you can now use

the “vlan” command under global configuration mode as well.

BR1#config t

Enter configuration commands, one per line. End with CNTL/Z.

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BR1(config)#int vlan 120

BR1(config-if)#ip address 10.1.120.1 255.255.255.0

BR1(config-if)#description Data

BR1(config-if)#int vlan 130

BR1(config-if)#ip address 10.1.130.1 255.255.255.0

BR1(config-if)#description Voice

BR1(config-if)#exit

BR1(config)#interface range FastEthernet 0/3/0 - 1

BR1(config-if-range)#switchport voice vlan 130

BR1(config-if-range)#switchport access vlan 120

BR1(config-if-range)#no shut

! Tip - Don’t forget to add the new networks into the routing protocol.

BR1(config)#router ospf 1

BR1(config-router)#network 10.1.120.0 0.0.0.255 area 0

BR1(config-router)#network 10.1.130.0 0.0.0.255 area 0

BR2:

BR2#vlan database

BR2(vlan)#vlan 220 name Data state active

VLAN 220 added:

Name: Data

State ACTIVE

BR2(vlan)#vlan 230 name Voice state active

VLAN 230 added:

Name: Voice

State ACTIVE

BR2(vlan)#exit

APPLY completed.

Exiting....

BR2#config t

Enter configuration commands, one per line. End with CNTL/Z.

BR2(config)#int vlan 220

BR2(config-if)#ip address 10.1.220.1 255.255.255.0

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BR2(config-if)#description Data

BR2(config-if)#int vlan 230

BR2(config-if)#ip address 10.1.230.1 255.255.255.0

BR2(config-if)#description Voice

BR2(config-if)#exit

BR2(config)#interface range FastEthernet 0/1/0 - 1

BR2(config-if-range)#switchport voice vlan 230

BR2(config-if-range)#switchport access vlan 220

BR2(config-if-range)#no shut

! Tip - Don’t forget to add the new networks into the routing protocol.

BR2(config)#router ospf 1

BR2(config-router)#network 10.1.220.0 0.0.0.255 area 0

BR2(config-router)#network 10.1.230.0 0.0.0.255 area 0

2. Validate your configurations with the “show

vlan” ( or “show current” under the vlan database)

and “show cdp neighbors” commands.

!

HQ-3750#show vlan

VLAN Name Status Ports

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

1 default active ! Fa1/0/1, Fa1/0/2, Fa1/0/3, Fa1/0/4,

! ! ! ! ! ! Fa1/0/5, Fa1/0/8, Fa1/0/9, Fa1/0/10,

! ! ! ! ! ! Fa1/0/11, Fa1/0/12, Fa1/0/13, Fa1/0/14,

! ! ! ! ! ! Fa1/0/15, Fa1/0/16, Fa1/0/17, Fa1/0/18,

! ! ! ! ! ! Fa1/0/19, Fa1/0/20, Fa1/0/21, Fa1/0/22,

! ! ! ! ! ! Fa1/0/23, Gi1/0/1, Gi1/0/2

20 Data active

30 Voice active

1002 fddi-default act/unsup

1003 token-ring-default act/unsup

1004 fddinet-default act/unsup

1005 trnet-default act/unsup

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HQ-3750#show cdp neigh

Capability Codes: R - Router, T - Trans Bridge, B - Source Route Bridge

S - Switch, H - Host, I - IGMP, r - Repeater, P - Phone

Device ID Local Intrfce Holdtme Capability Platform Port ID

CCMSUB Fas 1/0/1 152 H none foundeth0

CCMPUB Fas 1/0/1 157 H none foundeth0

HQ_Rtr Fas 1/0/24 140 R S I 2811 Fas 0/0.1

uccx. Fas 1/0/5 172 H Win2000 SeEth 1/1

presence Fas 1/0/4 145 H none foundeth0

SEP003094C27864 Fas 1/0/6 138 H P IP Phone 7Port 1

SEP003094C2DA38 Fas 1/0/7 132 H P IP Phone 7Port 1

SEP0002FDAEFDDF Fas 1/0/8 159 H P IP Phone 7Port 1

Unity Fas 1/0/3 162 H none foundeth0

PSTN-WAN Fas 1/0/22 124 R S I 2811 Fas 0/0

HQ-3750#

!Note - The publisher and subscriber are on the same physical box so they will

show as if they are connected to the same port on the switch. This is normal

and you should configure FastEthernet 1/0/2 as if the subscriber is plugged

into that port.

BR1#vlan database

BR1(vlan)#show current

VLAN ISL Id: 1

Name: default

Media Type: Ethernet

VLAN 802.10 Id: 100001

State: Operational

MTU: 1500

Translational Bridged VLAN: 1002

Translational Bridged VLAN: 1003

VLAN ISL Id: 120

Name: Data

Media Type: Ethernet

VLAN 802.10 Id: 100220

State: Operational

MTU: 1500

VLAN ISL Id: 130

Name: Voice

Media Type: Ethernet

VLAN 802.10 Id: 100230

State: Operational

MTU: 1500

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BR1#show cdp neigh

Capability Codes: R - Router, T - Trans Bridge, B - Source Route Bridge

S - Switch, H - Host, I - IGMP, r - Repeater

Device ID Local Intrfce Holdtme Capability Platform Port ID

HQ_Rtr Ser 0/1/0.1 161 R S I 2811 Ser 0/2/0.1

SEP0016C72D3DCC Fas 0/3/1 146 H IP Phone Port 1

SEP003094C3DCF9 Fas 0/3/0 143 H IP Phone Port 1

BR1#

BR2# show cdp neigh

Capability Codes: R - Router, T - Trans Bridge, B - Source Route Bridge

S - Switch, H - Host, I - IGMP, r - Repeater

Device ID Local Intrfce Holdtme Capability Platform Port ID

HQ_Rtr Ser 0/2/0.1 133 R S I 2811 Ser 0/2/0.2

SEP001F6C7FD6C1 Fas 0/1/1 167 H IP Phone Port 1

SEP001F6C7F722B Fas 0/1/0 167 H IP Phone Port 1

BR2#

BR2#vlan database

BR2(vlan)#show current

VLAN ISL Id: 1

Name: default

Media Type: Ethernet

VLAN 802.10 Id: 100001

State: Operational

MTU: 1500

Translational Bridged VLAN: 1002

Translational Bridged VLAN: 1003

VLAN ISL Id: 220

Name: Data

Media Type: Ethernet

VLAN 802.10 Id: 100220

State: Operational

MTU: 1500

VLAN ISL Id: 230

Name: Voice

Media Type: Ethernet

VLAN 802.10 Id: 100230

State: Operational

MTU: 1500

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3. In order to accomplish this we need to first create

the dhcp excluded address and then define the DHCP

pool:

HQ_Rtr(config)#ip dhcp excluded-address 10.1.30.1 10.1.30.119

HQ_Rtr(config)#ip dhcp excluded-address 10.1.30.131 10.1.30.255

HQ_Rtr(config)#ip dhcp pool IPPHONE

HQ_Rtr(config-config)#network 10.1.30.0 255.255.255.0

HQ_Rtr(config-config)#default-router 10.1.30.1

HQ_Rtr(config-config)#option 150 ip 10.1.200.21 10.1.200.25

Time Saver - In order to save time you should use notepad to copy and

paste configurations which may be the same or need little modification.

4. We can use the same configuration from the last task

but modify for BR1:

BR1(config)#ip dhcp excluded-address 10.1.130.1 10.1.130.129

BR1(config)#ip dhcp excluded-address 10.1.130.141 10.1.130.255

BR1(config)#ip dhcp pool IPPHONE

BR1(config-config)#network 10.1.130.0 255.255.255.0

BR1(config-config)#default-router 10.1.130.1

BR1(config-config)#option 150 ip 10.1.200.21 10.1.200.25

5. Perform the same for BR2:

BR2(config)#ip dhcp excluded-address 10.1.230.1 10.1.230.209

BR2(config)#ip dhcp excluded-address 10.1.230.221 10.1.230.255

BR2(config)#ip dhcp pool IPPHONE

BR2(config-config)#network 10.1.230.0 255.255.255.0

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BR2(config-config)#default-router 10.1.230.1

BR2(config-config)#option 150 ip 10.1.230.1

6. Next configure each router for the appropriate time

zone and NTP settings:

HQ_Rtr(config)#clock timezone PST -8

HQ_Rtr(config)#clock summer-time PDT recurring

HQ_Rtr(config)#ntp master 2

BR1(config)#clock timezone EST -5

BR1(config)#clock summer-time EDT recurring

BR1(config)#ntp server 10.10.32.1

BR2(config)#clock timezone UTC +5 30

BR2(config)#clock summer-time PDT recurring

BR2(config)#ntp server 10.10.32.1

! Tip - Use can use your candidate PC to look up the time zone of most

countries but most likely they will specify for you.

! Tip - Don’t forget to set the time zone settings in CallManager as it may not be

mentioned in the phone setup tasks.

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CallManager and CallManager Express:

1. This tasks requires a lot of configuration as this is

the base line for everything you do in CallManager. We

will setup everything from device pools, locations,

regions, media resource lists and groups, route lists

and route groups, partitions and calling search

spaces.

!

Lets start off with service activation first as

nothing will work if we don’t have services activated:

Make sure you enable services on both servers and all

services except the “Cisco Messaging Interface”.

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!Tip - Enabling all services as stated above will ensure that all needed services

are activated. Unless you are specifically told to only enable needed services

then this strategy will allow you to make sure all the required services are

running.

Next we need to start configuring the required

CallManager settings needed to allow for the

registration of HQ and BR1 phones.

Make sure the proper servers are defined:

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Next create a CallManager group that will allow for

devices to register to the subscriber and then the

publisher:

When finished you should see the following to groups:

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! Note - The requirement for registration order is normally on the first page of

instructions.

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Next we need to create the regions:

We need to make sure the settings between the regions

are set appropriately:

Time Saver - This is where reading through the whole lab will save a lot of

time as we can set the proper region settings and never return to this page.

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Now we need to create the HQ and BR1 Date/Time groups:

! Note - This is also where we could set the date/time format if asked during a

specific task.

Next we need to create the locations:

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Time Saver - This is where reading through the whole lab will save a lot of

time as we can set the proper location settings and never return to this page.

We now need to create the media resource groups and

lists:

!Tip - In reading the lab we know that there is a need to separate the MOH

servers and it is always a good idea to separate hardware from software

resources.

We can then create the needed media resource lists:

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Below is the full list of the media resource lists:

We now need to create the device pools:

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Time Saver - We know from reading the lab that one phone will not be

allowed to register in SRST. We can create the device pool for this function

now in order to save time later.

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Here is the final list of device pools:

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Next we need to check the Enterprise Parameters and

make changes as needed:

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!Tip - Changing the host name to an IP address in the URL settings will prevent

authentication and IP phone service failures. This is highly recommended

whether you have a DNS server or not!

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We can also create the AAR groups now as this will

help to cut down the number of times we need to visit

the phone configuration page:

Once we have both groups created we need to set the

dial prefix according to the dial plan:

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Now we will need to create the the partitions and

calling search spaces that will be needed for the lab:

On the next page is a table of calling search spaces

and the partitions that belong to each calling search

space.

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!"##$%&'()"*+,'(-"+)'

."/)' 0)1+*$-2$3%' 4352)'6"*2$2$3%1'73*'2,$1'!"##$%&'()"*+,'(-"+)'

!""#$%&#''%( !""#$%&#''%( )*+$%&+''%(

!""#$%&#,-*( !""#$%&#,-*( )*+$%&+.&&(

)*+$%&+,/012/34(

)*+$%&+*566$7)'""(

)*+89+,/012/34(

)*+$%:+,/012/34(

)*+$%&+6;<34(

)*+$%&+6=(

)*+$%&+,-*(

!""#$%&#,-*>%-'6( !""#$%&#,-*>%-'6( )*+$%&+.&&(

)*+$%&+,/012/34(

)*+$%&+*566$7)'""(

)*+89+,/012/34(

)*+$%:+,/012/34(

!""#$%&#6=( !""#$%&#6=( )*+$%&+.&&(

)*+$%&+,/012/34(

)*+$%&+*566$7)'""(

)*+89+,/012/34(

)*+$%:+,/012/34(

)*+$%&+6;<34(

)*+$%&+6=(

!""#$%&#65!'6( !""#$%&#65!'6( )*+$%&+.&&(

)*+$%&+,/012/34(

)*+$%&+*566$7)'""(

)*+89+,/012/34(

)*+$%:+,/012/34(

)*+$%&+6;<34(

!""#$%:( !""#$%:( )*+$%&+,/012/34(

)*+$%:+,/012/34(

)*+89+,/012/34(

)*+$%:+*566$7)'""(

!""#89#''%( !""#89#''%( )*+89+''%(

!""#89#,-*( !""#89#,-*( )*+89+.&&(

)*+89+,/012/34(

)*+89+*566$7)'""(

)*+$%&+,/012/34(

)*+$%:+,/012/34(

)*+89+6;<34(

)*+89+6=(

)*+89+,-*(

!""#89#,-*>%-'6( !""#89#,-*>%-'6( )*+89+.&&(

)*+89+,/012/34(

)*+89+*566$7)'""(

)*+$%&+,/012/34(

)*+$%:+,/012/34(

!""#89#6=( !""#89#6=( )*+89+.&&(

)*+89+,/012/34(

)*+89+*566$7)'""(

)*+$%&+,/012/34(

)*+$%:+,/012/34(

)*+89+6;<34(

)*+89+6=(

!""#89#65!'6( !""#89#65!'6( )*+89+.&&(

)*+89+,/012/34(

)*+89+*566$7)'""(

)*+$%&+,/012/34(

)*+$%:+,/012/34(

)*+89+6;<34(

!""#?5,!>@',6( !""#?5,!>@',6( )*+$%&+,/012/34(

)*+89+,/012/34(

(

(

We should now be able to add the phones per the

requirements in the table.

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HQ Phone 1:

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HQ Phone 2:

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BR1 Phone 1:

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BR1 Phone 2:

!Note - In reading the lab we know the BR1 Phone 2 requires another extension

“*2002”. We can create that as part of the initial phone setup which we have

done here or you could wait till that specific task.

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BR1 Phone 2 (Line 2):

! Tip - Most of the phone settings per phone are the same per site except for the

calling search space.

Time Saver - In order to save time you can use auto registration on the

callmanager and then modify each phone to the correct settings.

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We can now start the registration of phones within the

BR2 site. We will first need to define all the

required settings underneath the “telephony-service”

configuration. We can then create the phones and their

associated DN’s:

BR2(config)#telephony-service

BR2(config-telephony)#authentication credential admin cisco

BR2(config-telephony)#max-ephones 2

BR2(config-telephony)#max-dn 7

BR2(config-telephony)#system message Your current options

BR2(config-telephony)#ip source-address 10.1.230.1 port 2000

BR2(config-telephony)#load 7941 term41.default.loads

BR2(config-telephony)#voicemail 3180

BR2(config-telephony)#max-conferences 8 gain -6

BR2(config-telephony)#call-forward pattern .T

BR2(config-telephony)#dn-webedit

BR2(config-telephony)#time-webedit

BR2(config-telephony)#transfer-system full-blind

BR2(config-telephony)#transfer-pattern 9.T

BR2(config-telephony)#secondary-dialtone 9

BR2(config-telephony)#create cnf-files

BR2(config-telephony)#exit

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Now we can configure the phones and their DN’s:

BR2(config)#ephone-dn 1

BR2(config-ephone-dn)#number 3001

BR2(config-ephone-dn)#pickup-group 1

BR2(config-ephone-dn)#description 6303001

BR2(config-ephone-dn)#name BR2 Phone 1

BR2(config-ephone-dn)#call-forward noan 3180 timeout 5

BR2(config)#ephone-dn 2

BR2(config-ephone-dn)#number 3002

BR2(config-ephone-dn)#pickup-group 1

BR2(config-ephone-dn)#description 6303002

BR2(config-ephone-dn)#name BR2 Phone 2

We can also configure the extra DN’s specified for the

shared line on BR2 phone 1 and 2:

BR2(config)#ephone-dn 6 dual-line

BR2(config-ephone-dn)# number 3003

BR2(config-ephone-dn)# pickup-group 1

BR2(config-ephone-dn)# description 6303003

BR2(config)#ephone-dn 7 dual-line

BR2(config-ephone-dn)# number 3004

BR2(config-ephone-dn)# pickup-group 1

We can now configure the phones knowing that phone 1

will be using extension mobility:

BR2(config)#ephone 1

BR2(config-ephone)#conference drop-mode creator

BR2(config-ephone)#mac-address 001F.6C7F.722B

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BR2(config-ephone)#type 7941GE

BR2(config-ephone)#logout-profile 1

BR2(config)#ephone 2

BR2(config-ephone)#conference drop-mode creator

BR2(config-ephone)#mac-address 001F.6C7F.D6C1

BR2(config-ephone)#username "BR2Phone2"

BR2(config-ephone)#type 7941GE

BR2(config-ephone)#button 1:2 2o6,7

Next we need to allow the SCCP firmware to be

downloaded from the router flash:

BR2(config)#tftp-server flash:apps41.8-3-2-27.sbn BR2(config)#tftp-server flash:cnu41.8-3-2-27.sbn BR2(config)#tftp-server flash:cvm41sccp.8-3-2-27.sbn BR2(config)#tftp-server flash:dsp41.8-3-2-27.sbn BR2(config)#tftp-server flash:jar41sccp.8-3-2-27.sbn

!Tip - If you like to use the CME and CUE web interface the the following

commands should be entered at this stage to prepare you to use the web

interface.

BR2(config)#ip http server

BR2(config)#no ip http secure-server

BR2(config)#ip http path flash:

BR2(config)#telephony-service

BR2(config-telephony)#web admin system name admin password cisco

BR2(config-telephony)#url services http://10.1.230.2/voiceview/common/

login.do (Inserted by default in most cases)

BR2(config-telephony)#url authentication http://10.1.230.1/CCMCIP/

authenticate.asp (Inserted by default in most cases)

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2. This was already completed in the above solution with the

ephone-dn 6 and 7 configuration and the addition of this

button with overlay to phone 2 ephone. Adding this to

phone 1 will come later during the extension mobility

task.

3. This was already completed in the above solution with the

“conference drop-mode creator” command under the ephone

configuration for phone 2. Adding this to phone 1 will

come later during the extension mobility task.

4. This was accomplished in task 1 by setting the alerting

name, caller id, external phone number mask, and date/

time group format the same in CallManager.

5. This was accomplished in task 1 by setting setting the

“pickup-group 1” command under all the ephone-dn’s.

6. This is accomplished with the following configuration:

BR2(config)#ephone-hunt 1 sequential

BR2(config-ephone-hunt)#pilot 3111

BR2(config-ephone-hunt)#list 3001, 3003

BR2(config-ephone-hunt)#final 3000

BR2(config-ephone-hunt)#timeout 5, 5

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7. This is accomplished through creating a logout profile

and a user profile for phone 1:

BR2(config)#voice logout-profile 1

BR2(config-voice-profile)#pin 12345

BR2(config-voice-profile)#user br2phone3 password cisco

BR2(config-voice-profile)#number 3003,3004 type overlay

BR2(config)#voice user-profile 1

BR2(config-voice-profile)#pin 12345

BR2(config-voice-profile)#user BR2Phone1 password cisco

BR2(config-voice-profile)#number 3001 type normal

BR2(config-voice-profile)#number 3003,3004 type overlay

8. In this version of CUE and CME voice view is enabled by

default. One step that needs to take place is adding the

CME authentication URL as the backup URL for the voice

view service in CUE:

CUE(config)#service phone-authentication CUE(config-service)#fallback-url http://10.1.230.1/CCMCIP/

authenticate.asp CUE(config-service)#end phone-authentication

9. The is simple to accomplish by adding the following under

the ephone-dn:

BR2(config)#ephone-dn 2

BR2(config-ephone-dn)#mobility

BR2(config-ephone-dn)#snr 96305000 delay 5 timeout 10 cfwd-noan 3180

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!Note - Though the task asks us to allow the number to be reach by the full

E164 number we can perform this action later when setting up the call routing

as all calls will need to be striped to four digits.

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Voice Gateways and Signaling

1. In order to enable the HQ router as an MGCP gateway we

need the following configuration on the HQ router:

HQ_Rtr(config)#network-clock-participate wic 0

HQ_Rtr(config)#isdn switch-type primary-ni

HQ_Rtr(config)#voice-card 0

HQ_Rtr(config-voicecard)#dspfarm

HQ_Rtr(config-voicecard)#dsp services dspfarm

HQ_Rtr(config-voicecard)#exit

HQ_Rtr(config)#controller T1 0/0/0

HQ_Rtr(config-controller)#cablelength long 0db

HQ_Rtr(config-controller)#pri-group timeslots 1-3,24 service mgcp

HQ_Rtr(config-controller)#exit

HQ_Rtr(config)#interface Serial0/0/0:23

HQ_Rtr(config-if)# isdn bind-l3 ccm-manager

HQ_Rtr(config-if)#exit

HQ_Rtr(config)#ccm-manager redundant-host 10.1.200.21

HQ_Rtr(config)#ccm-manager mgcp

HQ_Rtr(config)#mgcp

HQ_Rtr(config)#mgcp call-agent 10.1.200.25 service-type mgcp version 0.1

HQ_Rtr(config)#mgcp fax t38 ecm

HQ_Rtr(config)#mgcp

! Note - In IOS release 12.4 the dial-peer no longer needs to be created with the

service command.

Time Saver - If you have multiple MGCP or H.323 gateways then use

notepad to copy and paste common configuration components.

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Next we will need to configure the gateway in

CallManager:

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!Tip - If you have set or been asked to set a domain name on the router the

device name for MGCP would be the router name plus the domain name. So in

our case it could be “HQ_Rtr.voiceie.com” if we had “voiceie.com” as a

domain name on the router.

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2. In order to enable the BR1 router as an H.323 gateway we

need the following configuration on the BR1 router:

BR1(config)#network-clock-participate wic 1

BR1(config)#isdn switch-type primary-ni

BR1(config)#voice-card 0

BR1(config-voicecard)#dsp services dspfarm

BR1(config-voicecard)#exit

BR1(config)#voice translation-rule 1

BR1(cfg-translation-rule)#rule 1 /\(^702333\)\(2...\)/ /\2/

BR1(cfg-translation-rule)#exit

BR1(config)#voice translation-profile STRIP

BR1(cfg-translation-profile)#translate called 1

BR1(cfg-translation-profile)#exit

BR1(config)#voice-port 0/1/0:23

BR1(config-voiceport)#translation-profile incoming STRIP

BR1(config-voiceport)#exit

BR1(config)#controller T1 0/1/0

BR1(config-controller)#framing esf

BR1(config-controller)# linecode b8zs

BR1(config-controller)# pri-group timeslots 1-3,24

BR1(config-controller)# exit

BR1(config)#interface Loopback0

BR1(config-if)#h323-gateway voip interface

BR1(config-if)#h323-gateway voip bind srcaddr 10.10.32.2

BR1(config-if)#exit

BR1(config)#interface Serial0/1/0:23

BR1(config-if)#isdn outgoing display-ie

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Next we will need to setup BR1 as an H.323 gateway in

CallManager:

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!Tip - The most common issue with H.323 gateways is that IP address specified

in CallManager is not the IP address that the H.323 process in the router is

binding. Make sure these IP address match!

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3. In order to enable the BR2 router as an H.323 gateway we

need the following configuration on the BR2 router:

BR2(config)#network-clock-participate wic 0

BR2(config)#isdn switch-type primary-net5

BR2(config)#voice translation-rule 1

BR2(cfg-translation-rule)#rule 1 /\(^44207630\)\(3...\)/ /\2/

BR2(cfg-translation-rule)#exit

BR2(config)#voice translation-profile STRIP

BR2(cfg-translation-profile)#translate called 1

BR2(cfg-translation-profile)#exit

BR2(config)#voice-card 0

BR2(config-voicecard)#dspfarm

BR2(config-voicecard)#dsp services dspfarm

BR2(config-voicecard)#exit

BR2(config)#controller E1 0/0/0

BR2(config-controller)#pri-group timeslots 1-3,16

BR2(config-controller)#exit

BR2(config)#interface Serial0/0/0:15

BR2(config-if)#isdn outgoing display-ie

BR2(config-if)#exit

BR2(config)#voice-port 0/0/0:15

BR2(config-voiceport)#translation-profile incoming STRIP

4. In order to setup the gatekeeper we need to perform the

following configuration on the HQ router:

HQ_Rtr(config)#gatekeeper

HQ_Rtr(config-gk)#zone local CM ccbootcamp.com 10.10.32.1

HQ_Rtr(config-gk)#gw-type-prefix 1#* default-technology

HQ_Rtr(config-gk)#no shutdown

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Next we need to enable CallManager to register to the

gatekeeper through a H.225 gatekeeper controlled

trunk. We can accomplish this by adding the gatekeeper

and then the trunk in CallManager:

!Tip - If you would like to lower the time it takes for gatekeeper to register

after a loss of connectivity then change the “Registration Retry Timeout” to 30

seconds.

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Now we need to add the trunk:

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! Tip - You can validate registration with the “show gatekeeper endpoints”

command on the HQ router.

HQ_Rtr#show gatekeeper endpoints

GATEKEEPER ENDPOINT REGISTRATION

================================

CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags

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

10.1.200.21 33072 10.1.200.21 32794 CM H323-GW

H323-ID: BR2_1

Voice Capacity Max.= Avail.= Current.= 0

10.1.200.25 33077 10.1.200.25 32785 CM H323-GW

H323-ID: BR2_2

Voice Capacity Max.= Avail.= Current.= 0

Total number of active registrations = 4

HQ_Rtr#

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Next we will need to configure the BR2 router is

register with the gatekeeper. This requires the

following configuration:

BR2(config)#interface Loopback0

BR2(config-if)#h323-gateway voip interface

BR2(config-if)#h323-gateway voip id CME ipaddr 10.10.32.1 1719

BR2(config-if)#h323-gateway voip h323-id CM

BR2(config-if)#h323-gateway voip tech-prefix 1#

BR2(config-if)#h323-gateway voip bind srcaddr 10.10.32.3

BR2(config-if)#exit

BR2(config)#gateway

! Tip - Once again you can verify with the “show gatekeeper endpoints”

command on the HQ router.

HQ_Rtr#show gatekeeper endpoints

GATEKEEPER ENDPOINT REGISTRATION

================================

CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags

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

10.10.32.3 1720 10.10.32.3 57205 CM H323-GW

H323-ID: CME

Voice Capacity Max.= Avail.= Current.= 0

10.1.200.21 33072 10.1.200.21 32794 CM H323-GW

H323-ID: BR2_1

Voice Capacity Max.= Avail.= Current.= 0

10.1.200.25 33077 10.1.200.25 32785 CM H323-GW

H323-ID: BR2_2

Voice Capacity Max.= Avail.= Current.= 0

Total number of active registrations = 4

HQ_Rtr#

!Note - You will notice that the directory numbers from the SIP CME did not

register with gatekeeper. This is to be expected as SIP CME phones cannot be

auto registered to gatekeeper.

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Call Routing

1. The has already been accomplished for HQ and BR1 when

we setup the phones previously by giving them the

proper calling search space. To enable class of

service for the BR2 site we need to create the

following COR list:

dial-peer cor custom

name Local

name LD

name INT

name Emergency

dial-peer cor list Emergency

member Emergency

dial-peer cor list Local

member Local

dial-peer cor list Int

member INT

dial-peer cor list LD

member LD

dial-peer cor list Phone1

member Local

member LD

member INT

member Emergency

dial-peer cor list Phone2

member Local

member LD

member Emergency

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Now that we have created the COR list we need to

assign the proper COR list to the voice register pool

of the proper phone:

BR2(config)#ephone-dn 1

BR2(config-ephone-dn)#cor incoming Phone1

BR2(config)#ephone-dn 2

BR2(config-ephone-dn)#cor incoming Phone2

2. In order to accomplish the next two tasks we need to

create all the route groups and route lists first. We

will first need to create the route groups:

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Then next the route lists in the following tables with

the proper settings:

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Page 653: Ccie Voice Advanced Lab Workbook

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Page 654: Ccie Voice Advanced Lab Workbook

A summary of all the route lists is below:

We can now create all the route patterns for the HQ

call routing and point them to the proper route list:

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! Note - All the digit manipulation and phone number masking is done in the

route lists and therefore should not be done in the route pattern.

Next we need to add the proper call routing to the

gatekeeper on the HQ gateway. Remember the CME will

automatically register it’s DN’s to the gatekeeper:

HQ_Rtr(config-gk)#zone prefix CM 7....

3. We can now use the previous created route groups and

route lists to setup the BR1 call routing by adding

the following route patterns:

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Page 657: Ccie Voice Advanced Lab Workbook

Now since the BR1 gateway is an H.323 gateway we need

to add the proper dial-peers to the BR1 gateway to

complete the call routing. We may also want to take

into account the SRST task for maintaining class of

service and create the COR list at the same time:

dial-peer cor custom

name Local

name LD

name INT

name Emergency

dial-peer cor list Emergency

member Emergency

dial-peer cor list Local

member Local

dial-peer cor list Int

member INT

dial-peer cor list LD

member LD

dial-peer cor list Phone1

member Local

member LD

member INT

member Emergency

dial-peer cor list Phone2

member Local

member LD

member Emergency

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Page 659: Ccie Voice Advanced Lab Workbook

We can now create the dial-peers with the COR already

configured:

dial-peer voice 1 pots

incoming called-number .

direct-inward-dial

dial-peer voice 2000 voip

preference 1

destination-pattern 2...

session target ipv4:10.1.200.25

ip qos dscp cs3 signaling

dial-peer voice 2001 voip

preference 2

destination-pattern 2...

session target ipv4:10.1.200.21

ip qos dscp cs3 signaling

dial-peer voice 7 pots

corlist outgoing Local

destination-pattern 9[2-9]......

port 0/1/0:23

forward-digits 7

dial-peer voice 11 pots

corlist outgoing LD

destination-pattern 91[2-9]..[2-9]......

port 0/1/0:23

forward-digits 11

!

dial-peer voice 911 pots

corlist outgoing Emergency

destination-pattern 911

port 0/1/0:23

forward-digits all

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!

dial-peer voice 9911 pots

corlist outgoing Emergency

destination-pattern 9911

port 0/1/0:23

forward-digits 3

!

dial-peer voice 9011 pots

corlist outgoing Int

destination-pattern 9011T

port 0/1/0:23

prefix 011

4. We can now setup the call routing for BR2 with the

following configuration on the BR2 router. We first

need to allow all the communications that maybe needed

in our environment:

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

supplementary-service h450.12

no supplementary-service sip refer

fax protocol cisco

h323

sip

registrar server expires max 1200 min 300

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Now we can create all the needed dial-peers with the

appropriate COR list which were previously defined:

dial-peer voice 1 pots

incoming called-number .

direct-inward-dial

!

dial-peer voice 2 pots

corlist outgoing Local

destination-pattern 90[1-9].......

port 0/0/0:15

prefix 0

!

dial-peer voice 1000 voip

preference 1

destination-pattern 71...

session target ras

!

dial-peer voice 1001 pots

preference 2

destination-pattern 71...

port 0/0/0:15

prefix 0017029461

!

dial-peer voice 2000 voip

preference 1

destination-pattern 72...

session target ras

!

dial-peer voice 2001 pots

preference 2

destination-pattern 72...

port 0/0/0:15

prefix 0017023332

!

dial-peer voice 7 pots

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corlist outgoing Local

destination-pattern 9[1-9]......

port 0/0/0:15

forward-digits 7

!

dial-peer voice 999 pots

corlist outgoing Emergency

destination-pattern 999

port 0/0/0:15

forward-digits all

!

dial-peer voice 900 pots

corlist outgoing Int

destination-pattern 900.T

port 0/0/0:15

prefix 00

We also need to add a translation pattern to strip the

“7” as the call comes into callmanager from BR2:

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We will also need to enable transcoding on the BR2

gateway to support calls transgressing from HQ/BR1 to

BR2:

sccp local FastEthernet0/0.230

sccp ccm 10.1.230.1 identifier 1 priority 1 version 3.1

sccp

sccp ccm group 1

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associate ccm 1 priority 1

associate profile 1 register TRANSCODE

dspfarm profile 1 transcode

codec g711ulaw

codec g711alaw

codec g729r8

codec g729abr8

codec g729ar8

codec g729br8

maximum sessions 4

associate application SCCP

telephony-service

sdspfarm units 2

sdspfarm transcode sessions 4

sdspfarm tag 1 TRANSCODE

!Tip - At this point you can start testing your calls to make sure they all

complete correctly. In order to test redundancy you can shutdown the

gatekeeper or the corresponding T1/E1 controller to test.

Time Saver - By saving the testing until your call routing is complete you

can save time by not interrupting your configuration and then fixing as you

test.

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Codec and Call Admission Control

1. This task was already accomplished earlier when we

created the regions.

2. This task was already accomplished earlier when we

created the regions.

3. This task was completed when we create the locations

earlier.

4. In order to accomplish this task we need to enable via

zone (IP-to-IP) control on the gatekeeper:

HQ_Rtr(config)#gatekeeper

HQ_Rtr(config-gk)#zone local VIA ccbootcamp.com

HQ_Rtr(config-gk)#zone local CM ccbootcamp.com 10.10.32.1 invia VIA

outvia VIA enable-intrazone

We then need to configure the HQ gateway for CUBE

functions:

voice service voip

allow-connections h323 to h323

interface Loopback0

h323-gateway voip interface

h323-gateway voip id VIA ipaddr 10.10.32.1 1719

h323-gateway voip h323-id IPIP

h323-gateway voip tech-prefix 1#

h323-gateway voip bind srcaddr 10.10.32.1

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gateway

In order to support the transcoding for the IP to IP

gateway we need to configure another transcoding

resource for use by the IP-to-IP gateway on the HQ

router:

telephony-service

sdspfarm units 1

sdspfarm transcode sessions 2

sdspfarm tag 1 IPIPTRANS

max-ephones 1

max-dn 1

ip source-address 10.10.32.1 port 2000

max-conferences 8 gain -6

transfer-system full-consult

create cnf-files

dspfarm profile 3 transcode

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

maximum sessions 2

associate application SCCP

sccp ccm 10.10.32.1 identifier 3 priority 3 version 3.1

sccp ccm group 2

associate ccm 3 priority 3

associate profile 3 register IPIPTRANS

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Next we need to create the dial-peers for the IP-to-IP

gateway on the HQ router:

dial-peer voice 1000 voip

destination-pattern 7....

session target ras

codec g711ulaw

!

dial-peer voice 1001 voip

destination-pattern 3...

session target ras

We can then change the region settings for HQ to BR2

to be the G7ll codec:

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!Tip - There is one troubleshooting problem I have thrown into the mix. If you

try and call from BR2 to HQ or BR1 the call may or may not fail. This is due

to the order of the partitions under the CSS_BR2 calling search space. You will

need to move the PT-BR2-Internal to the top of the list of partitions in order to

make all calls work correctly.

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High Availability Features

1. In order to accomplish this task we need to enable

“call-manager-fallback” in the BR1 gateway:

application

global

service alternate DEFAULT

call-manager-fallback

secondary-dialtone 9

max-conferences 4 gain -6

ip source-address 10.1.130.1 port 2000

max-ephones 2

max-dn 10

transfer-pattern .T

voicemail 917029461170

call-forward pattern .T

2. This task was accomplished earlier by setting the BR1

phone 1 device pool to “BR1-NOSRST” which had the SRST

reference disabled.

3. This was completed during the call routing section as

the dial-peers were created with 9 at the beginning.

4. In order to accomplish this we need to add the proper

COR list to the DN of BR1 phone 2 in the incoming

direction:

call-manager-fallback

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cor incoming Phone2 1 2002

5. Most of this task has already been completed during

the phone setup section. We created the AAR groups

with the appropriate prefixes and then assigned the

DN’s to the appropriate AAR group. The last step is to

enable the AAR service via the callmanager service

parameters:

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Media Resources Management

1. In order to accomplish this task we first need to

create the resource in callmanager:

Next we need to configure the SCCP and DSPFARM

settings on the HQ router:

sccp local Loopback0

sccp ccm 10.1.200.21 identifier 2 priority 2 version 7.0

sccp ccm 10.1.200.25 identifier 1 priority 1 version 7.0

sccp

!

sccp ccm group 1

associate ccm 1 priority 1

associate ccm 2 priority 2

associate profile 2 register HQTRANS

associate profile 1 register HQCONF

!

dspfarm profile 2 transcode

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codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

maximum sessions 4

associate application SCCP

!

dspfarm profile 1 conference

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

codec g729r8

codec g729br8

maximum sessions 4

associate application SCCP

!Tip - You can use the “show sccp” and the show “dspfarm profile” command

to verify gateway configuration. You may need to disable and then enable the

SCCP function if the resources aren’t registering.

2. In order to accomplish this task we first need to

create the resource in callmanager:

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Time Saver - This is another great instance of where copy and paste can

help you as the configuration will be very similar to the HQ router.

sccp local Loopback0

sccp ccm 10.1.200.25 identifier 1 priority 1 version 7.0

sccp ccm 10.1.200.21 identifier 2 priority 2 version 7.0

sccp

!

sccp ccm group 1

associate ccm 1 priority 1

associate ccm 2 priority 2

associate profile 1 register BR1CONF

associate profile 2 register BR1TRANS

!

dspfarm profile 2 transcode

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

codec gsmfr

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maximum sessions 4

associate application SCCP

!

dspfarm profile 1 conference

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

codec g729r8

codec g729br8

maximum sessions 4

associate application SCCP

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3. To accomplish this task we first need to create the

resource in callmanager:

The gateway configuration portion was accomplished

above in the conference bridge section.

4. To accomplish this task we first need to create the

resource in callmanager:

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The gateway configuration portion was accomplished

above in the conference bridge section.

5. In a previous step we create all the media resource

groups and list. Now we can add the proper media

resources to the proper groups. The publisher MOH

server will need to be put into the “PUB_MOH” media

resource group and we will need to make sure that this

is the MOH only server listed in the media resource

list for HQ.

6. We need to perform the same actions for BR1 but with

the subscriber server being put into the “SUB_MOH”

media resource group and making sure that this is the

only MOH server listed in the media resource list for

BR1. In addition we will need to enable the default

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audio source, subscriber MOH server, and the “SUB_MOH”

media resource group for multicasting:

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We need to insure that the “Max Hops” is set to “1” so

that the multicast stream stays local.

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Now we need to enable the G.729 codec through the

Cisco IP Voice Media Streaming App service parameters:

We then need to configure the BR1 gateway to support

multicast and the router to play the MOH file from

flash:

call-manager-fallback

moh music-on-hold.au

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multicast moh 239.1.1.1 port 16388 route 10.1.130.1

ip multicast-routing

interface Vlan130

ip pim sparse-dense-mode

! Tip - To test place a call from HQ to BR1 and then put the call on hold from

both ends. You should receive MOH but hear different music on both sides.

7. In order to accomplish this we first need to create

the MeetMe conference resources in callmanager:

We then need to change the callmanager service

parameter to allow for 10 participants per conference:

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8. This is accomplished through the same callmanager

service parameter section as the task above by setting

the “Drop Ad Hoc Conference” parameter correctly.

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QoS Features

1. We can accomplish this with the following

configuration.

On the HQ gateway:

interface Serial0/2/0.1 point-to-point

bandwidth 384

frame-relay interface-dlci 201

auto qos voip trust fr-atm

On the BR1 Gateway:

interface Serial0/2/0.1 point-to-point

bandwidth 384

frame-relay interface-dlci 101 ppp

auto qos voip trust fr-atm

The on both gateways modify the auto generated policy

map per the requirements:

policy-map AutoQoS-Policy-Trust

class AutoQoS-VoIP-RTP-Trust

priority percent 33

class AutoQoS-VoIP-Control-Trust

bandwidth percent 5

class class-default

fair-queue

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2. We can accomplish this with the following

configuration.

On the HQ gateway:

interface Serial0/2/0.2 point-to-point

bandwidth 1200

frame-relay interface-dlci 202

auto qos voip trust

On the BR2 gateway:

interface Serial0/2/0.1 point-to-point

bandwidth 1200

frame-relay interface-dlci 102

auto qos voip trust

Then on both gateways modify the auto generated policy

map per the requirements:

policy-map AutoQoS-Policy-Trust2

class AutoQoS-VoIP-RTP-Trust

priority 360

class AutoQoS-VoIP-Control-Trust

bandwidth 60

class class-default

fair-queue

3. This was accomplished earlier under the two dial-peers

pointing traffic back to the callmanagers.

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Application and End Users

In order to move forward past this section we need to

define all the users. Outlined below in tables are the

users that need to be created and the settings for those

users.

!Tip - Pay close attention to the associated devices and rights of each user.

Before you start to add user you will need to create a “Standard AXL User”

group and assign the “Standard AXL API Access” role to that group.

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!""#$%&'$()*+,-.*

+,-.*/0* /)1(.2&'$()*

!"#$%&&

'(()#*+"#,-&./01&2-3,14+"#,-&

510/0-*0&$1,6(& 7"+-8+18&510/0-*0&91,6(&

'**0("&510/0-*0&76:/*1#("#,-& ;&

'**0("&<6"=,3=8#+),9&>?@?>& ;&

'**0("&.-/,)#*#"08&;,"#3#*+"#,-& ;&

'**0("&>0()+*0/&A0+801& ;&

B0C#*0&2-3,14+"#,-&

!,-"1,))08&B0C#*0/& 7?5DDEF!GHBIB!!&7?5DDIDJK!HGLFK&7?5DDIDJK!HB'IL&7?5DDIDJK!IB!@J&

!'5@&2-3,14+"#,-&

'//,*#+"08&!'5@&51,3#)0/& &

5014#//#,-/&2-3,14+"#,-&

$1,6(/& 7"+-8+18&!M2&')),%&!,-"1,)&,3&'))&B0C#*0/&7"+-8+18&!M2&?-+:)08&

>,)0/& 7"+-8+18&!M2&')),%&!,-"1,)&,3&'))&B0C#*0/&7"+-8+18&!M2&?-+:)08&

&&

25!!&&

'(()#*+"#,-&./01&2-3,14+"#,-&

510/0-*0&$1,6(& 7"+-8+18&510/0-*0&91,6(&

'**0("&510/0-*0&76:/*1#("#,-& ;&

'**0("&<6"=,3=8#+),9&>?@?>& ;&

'**0("&.-/,)#*#"08&;,"#3#*+"#,-& ;&

'**0("&>0()+*0/&A0+801& ;&

B0C#*0&2-3,14+"#,-&

!,-"1,))08&B0C#*0/& &

!'5@&2-3,14+"#,-&

'//,*#+"08&!'5@&51,3#)0/& &

5014#//#,-/&2-3,14+"#,-&

$1,6(/& 7"+-8+18&'NO&./01&

>,)0/& 7"+-8+18&'NO&'52&'**0//&

&&

.-#"P&&

'(()#*+"#,-&./01&2-3,14+"#,-&

510/0-*0&$1,6(& 7"+-8+18&510/0-*0&91,6(&

'**0("&510/0-*0&76:/*1#("#,-& ;&

'**0("&<6"=,3=8#+),9&>?@?>& ;&

'**0("&.-/,)#*#"08&;,"#3#*+"#,-& ;&

'**0("&>0()+*0/&A0+801& ;&

B0C#*0&2-3,14+"#,-&

!,-"1,))08&B0C#*0/& &

!'5@&2-3,14+"#,-&

'//,*#+"08&!'5@&51,3#)0/& &

5014#//#,-/&2-3,14+"#,-&

$1,6(/& 7"+-8+18&'NO&./01&

>,)0/& 7"+-8+18&'NO&'52&'**0//&

&

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The next list of Application users is created during the

setup of the contact center server and therefore will not

need to be create:

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!""#$%&'$()*+,-.*

+,-.*/0* /)1(.2&'$()*

!

"#$%&'('!!

)%%*&+$#&,-!./01!2-3,14$#&,-!

510/0-+0!61,7%! 8#$-9$19!510/0-+0!:1,7%!

)++0%#!510/0-+0!87;/+1&%#&,-! <!

)++0%#!=7#>,3>9&$*,:!?@A@?! <!

)++0%#!.-/,*&+&#09!<,#&3&+$#&,-! <!

)++0%#!?0%*$+0/!B0$901! <!

C0D&+0!2-3,14$#&,-!

E,-#1,**09!C0D&+0/! ))('F'G!))('F'H!))('F'I!))('F'J!))('F'K!))!2EC!

E)5A!2-3,14$#&,-!

)//,+&$#09!E)5A!51,3&*0/! !

5014&//&,-/!2-3,14$#&,-!

61,7%/! 8#$-9$19!EL2!@-$;*09!

?,*0/! 8#$-9$19!EL2!@-$;*09!

!!

14"#$%&'!!

)%%*&+$#&,-!./01!2-3,14$#&,-!

510/0-+0!61,7%! 8#$-9$19!510/0-+0!:1,7%!

)++0%#!510/0-+0!87;/+1&%#&,-! <!

)++0%#!=7#>,3>9&$*,:!?@A@?! <!

)++0%#!.-/,*&+&#09!<,#&3&+$#&,-! <!

)++0%#!?0%*$+0/!B0$901! <!

C0D&+0!2-3,14$#&,-!

E,-#1,**09!C0D&+0/! 8@5FF'HEIMCNCEE!8@5FFNFKOEMIJHO!8@5FFNFKOEMC)NJ!8@5FFNFKOENCEAK!

E)5A!2-3,14$#&,-!

)//,+&$#09!E)5A!51,3&*0/! !

5014&//&,-/!2-3,14$#&,-!

61,7%/! 8#$-9$19!EL2!@-$;*09!

?,*0/! 8#$-9$19!EL2!@-$;*09!

!!

!

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The following End Users will also need to be created with

the correct settings from the table:

!"#$%&'($)*+"#'"&'#,$

%&'($-.$ /'0'12+"'$

3456'($

*+"7(+00'#$

.'89:'&$

*+"7(+00'#$

;(+<90'&$

=&&+:9>7'#$

!?7'"&9+"&$

@(+41&$ A+0'&$

!"#$ %&&#$ '()&&*&+,-*.-/+$ $$ %&&#$01$)23!"#341567189$%&&#$3$4)--$

'581:87:$--;$(1:$<=67=$'581:87:$-24$(18>96:$

'581:87:$--;$(1:$<=67=$'581:87:$--;<'("$?:@010=57850A1$'581:87:$-24$(18>96:$

!"%$ %&&%$ '()&&#B-C%.*.--$ $$ %&&%$01$)23!"#341567189$

'581:87:$--;$(1:$<=67=$'581:87:$-24$(18>96:$

'581:87:$--;$(1:$<=67=$'581:87:$--;<'("$?:@010=57850A1$'581:87:$-24$(18>96:$

DE#$ #&&#$ '()&&*&+,-%CFB,$<)-DE#$

$$ #&&#$01$)23DE341567189$#&&#$3$4)--$

'581:87:$--;$(1:$<=67=$'581:87:$-24$(18>96:$

'581:87:$--;$(1:$<=67=$'581:87:$--;<'("$?:@010=57850A1$'581:87:$-24$(18>96:$

DE%$ #&&%$ '()&&*&+,-%.?*F$ $$ #&&%$01$)23DE341567189$

'581:87:$--;$(1:$<=67=$'581:87:$-24$(18>96:$

'581:87:$--;$(1:$<=67=$'581:87:$--;<'("$?:@010=57850A1$'581:87:$-24$(18>96:$

G7=8:@01$ $$ $$ $$ H$IA16$J$ $$ $$

KAL1$ #&&C$ $$ MAL1$ #&&C$01$)23DE341567189$

'581:87:$--;$(1:$<=67=$'581:87:$-24$(18>96:$

'581:87:$--;$(1:$<=67=$'581:87:$--;<'("$?:@010=57850A1$'581:87:$-24$(18>96:$

!

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Voicemail Integrations

1. In order to accomplish this we first need to perform

all the required setup in callmanager for voicemail.

We start first with creating the voicemail ports:

Time Saver - You can create the ports manually or through the wizard.

Obviously the wizard is designed to save time as it creates the line group

for you will all the voicemail ports as members.

Below is the line group configuration:

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Next you will need to add this line group to a hunt list:

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Now we need to create the voicemail pilot (Hunt Pilot)

and point it to the correct hunt list:

We can now create the MWI on and MWI off extensions:

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We can now set the default voicemail pilot and

profile:

!Tip - You will see that we create a profile named strip. This will be used later

in task 5 in order to strip the “*” out of the extension before it rolls to

voicemail. The settings of this profile are shown below.

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Now in order to start the integration with Unity

Connections we will first need to create a port group:

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Next will need to define the callmanager servers Unity

will use through the Edit->Servers page under the port

group:

During the port group creation we were asked how many

ports and the device name prefix. This created the

ports for us automatically:

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Now we can finish the integration by settings the

parameters for the phone system to include the AXL

servers for importing users from callmanager:

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! Tip - This is also where the loop detection is enabled which is required in the

next task.

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Now we can define the AXL servers through the Edit->

Cisco Unified Communications Manager AXl Servers under

the phone system:

!Note - You will see the application user “Unity” which we create previously.

This user must be in the “Standard AXL User Group” that we created in order

to import users from callmanager.

2. In Unity Connections loop prevention is on by default.

However, you should be aware that there are two ways to

accomplish loop prevention. One is by extension and the

other is through DTMF tone based recognition.

3. In order to accomplish this task you must first modify

the default voicemail user template to change the default

password. You can then import all the users into Unity:

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You can now go to the import users page an import all

users:

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When finished your users page should look like the

following:

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! Tip - If the primary extension of the user is not set then the user will not show

up for import.

4. In order to configure the auto attendant we will first

need to create the appropriate CTI route point in

callmanager and forward all calls to voicemail for this

CTI route point:

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Time Saver -You can see two CTI route points listed as the second one will

be used to record the greeting for the call handler in this task.

We can setup the Greetings Administrator access for HQ

phone 1 user through the call routing section:

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Once this is complete we can now setup the call

handler for the auto attendant:

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Next we can setup the caller input options as required

in the task:

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We now need to enable and setup the closed extension

to take a message:

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We also need to modify the standard greeting so that a

message is not taken:

We need to also assign the call handler owner so that

the HQ phone 1 user can access the call handler and

change the greeting via the Greetings Administrator:

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5. Part of this task has already been configured during the

phone setup section and callmanager voicemail section.

What we need to double check is that the voicemail

profile named “STRIP” is assigned to the BR1 phone 2

extension of “*2002”:

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6. In order to accomplish this task we first need to define

access to the CUE module, the ephone-dn’s for MWI, and

then dial-peers for the correct number(s) that need to be

forwarded to CUE:

interface Service-Engine1/0

ip unnumbered FastEthernet0/0.230

service-module ip address 10.1.230.2 255.255.255.0

service-module ip default-gateway 10.1.230.1

ip route 10.1.230.2 255.255.255.255 Service-Engine1/0

ephone-dn 4

number 8001....

mwi on

ephone-dn 5

number 8002....

mwi off

dial-peer voice 3180 voip

destination-pattern 3180

session protocol sipv2

session target ipv4:10.1.230.2

dtmf-relay rtp-nte

codec g711ulaw

no vad

!

dial-peer voice 3000 voip

destination-pattern 3000

session protocol sipv2

session target ipv4:10.1.230.2

dtmf-relay rtp-nte

codec g711ulaw

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no vad

Time Saver - In order to save time we can go ahed and create the dial-peer

need for task 8.

You can now start the CUE initialization process by

session into the CUE module from the BR2 gateway with

the following command:

BR2#service-module service-engine 1/0 session

Once you have completed the setup the CUE module will

reboot and is ready for configuration via the web

interface or via command line. You will now need to

set the MWI on and MWI off:

ccn application ciscomwiapplication aa description "ciscomwiapplication" enabled maxsessions 8 script "setmwi.aef" parameter "CallControlGroupID" "0" parameter "strMWI_OFF_DN" "8002" parameter "strMWI_ON_DN" "8001" end application

You can test access by pressing the voicemail button

on any BR2 phone.

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7. You can accomplish this via the web interface or via the

command line. We will show through the command line:

username br2phone3 create username BR2Phone2 create username BR2Phone1 create username admin create

username BR2Phone2 phonenumber "3002"

username BR2Phone1 phonenumber "3001"

username BR2Phone2 phonenumberE164 "442076303002"

username BR2Phone1 phonenumberE164 "442076303001"

voicemail mailbox owner "BR2Phone1" size 21176 description "BR2Phone1 mailbox" end mailbox voicemail mailbox owner "BR2Phone2" size 21176 description "BR2Phone2 mailbox" end mailbox

! Tip - You can test access and MWI by leaving a message and making sure the

MWI is working for each mailbox.

8. Part of this task was completed in task 6 and now we just

need to define the auto attendant in CUE:

ccn trigger sip phonenumber 3000

application "autoattendant"

enabled

maxsessions 8

end trigger

9. The was already accomplished earlier during the call

routing section. You will just need to test!

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Page 717: Ccie Voice Advanced Lab Workbook

IPCC Express Voice Application

1. In order to accomplish this task you will need to

complete the normal IPCC Express setup. Once this is

done you will need to create a Call Control Group

based on the task requirements:

!Tip - If required you can click the “Show More...” button and set the partition,

calling search space, AAR group, and other settings that may be required. In

this case we would probably want to set things like the device pool and

location.

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2. To accomplish this task you will need to create a new

application and trigger using the default auto

attendant script:

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3. First we need to create the application and trigger:

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We now can modify the resources and assign the skills

required for task 5:

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! Tip - If you don’t see any resources in IPCC Express then you probably

haven’t assigned the users an IPCC extension in callmanager.

We now need to create the contact services queue with

the same name we referenced under the application:

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Page 724: Ccie Voice Advanced Lab Workbook

We now need to create the IP Agent Phone Service:

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We can then assign the IP phone service to the proper

phones:

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You should now be able to login to the IP Agent Phone

service and test calls into the queue.

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4. In order to accomplish this we can modify the default

script to place the call on hold and then off hold

during the queuing process:

5. This was accomplished in task 3 by assigning the users

skills and then modifying the queue to route based on

“Most Skilled”.

6. This is shown in the script example provided under

task 4.

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! Tip - You should get to know the “Get Reporting Statistics” function in IPCC

Express. The function will allow you to create checks for just about any task.

CallManager Voice Application

1. Part of this task was already completed when we created

the user in callmanager. We now need to create the IP

phone service:

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Page 729: Ccie Voice Advanced Lab Workbook

Next we can create the device profile:

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We need to make sure we subscribe the device profile

to the extension mobility phone service:

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We then need to enable the required phones for extension

mobility and subscribe them to the IP phone service for

extension mobility:

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Page 732: Ccie Voice Advanced Lab Workbook

Presence

1. The first step is to run through the setup for

Presence after the initial install. Once this is

complete we can setup the required SIP trunk in

callmanager and set the SIP publish trunk in the

callmanager service parameters:

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Now set the CUP Publish Trunk in the callmanager

service parameters:

We can now create the CUPC in callmanager:

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Make sure on all the phones and CUPC in callmanager

that the subscribe calling search space is set

accordingly:

Next we need to assign the capabilities in

callmanager:

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Once this is complete we now need to finish the

configuration of the presence server. We can start by

validating the setup via System->CUCM Publisher:

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You can now configure the incoming and outgoing ACL’s:

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We now need to configure the CM SIP Publish trunk

under the Presence->Settings:

We now need to configure the callmanager as a presence

gateway:

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Page 738: Ccie Voice Advanced Lab Workbook

We now need to modify the proxy services settings:

We can then start configuring the CUPC settings:

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Page 739: Ccie Voice Advanced Lab Workbook

Next we need to configure the Unity server as a

voicemail server and then configure the voicemail

profile:

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Page 741: Ccie Voice Advanced Lab Workbook

We can now set the proper profiles for the HQ phone 1

user:

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Page 742: Ccie Voice Advanced Lab Workbook

Next we need to configure the “Deskphone Control

Settings” so that the CUPC client can control the

phone:

! Tip - The phones that need to be controlled by the “CtiGw” user were defined

earlier during the Application User section.

! Tip - You will also want to make sure that the phone being controlled has an

owner assigned and is associated to the end user.

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Page 743: Ccie Voice Advanced Lab Workbook

We can then enable the HQ phone 1 user for desktop

phone control:

Next you will need to the proxy domain and restart the

presence and proxy services:

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Page 744: Ccie Voice Advanced Lab Workbook

!Tip - You will need to create a host file entry on the machine running CUPC

based on the domain name you select in the presence server parameters.

Example: “presence.ccbootcamp.com” pointing to 10.1.200.23.

2. We need to first enable Unified Client access within

Unity Connections:

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Page 745: Ccie Voice Advanced Lab Workbook

Next we need to setup the Unity account user name and

password in the CUPC client and check that all

services are up and running:

!Note - You can switch between modes and check services for each setup. Keep

in mind that the service for the mode your not in will show as down or not

configured.

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Chapter 5 - Lab 5

Brief Lab Overview:

This lab is designed to test your IPCC Express knowledge.

Basic Campus Design:

1. Configure VLANS and IP networks in all three sites

according to the following two tables:

!

VLAN HQ BR1 BR2

Servers 1 N/A N/A

Data 20 120 220

Voice 30 130 230

!

Network HQ BR1 BR2

Servers 10.1.200.0/24 N/A N/A

Data 10.YY.20.0/24 10.YY.120.0/24 10.YY.220.0/24

Voice 10.YY.30.0/24 10.YY.130.0/24 10.YY.230.0/24

2. Make sure you set all voice traffic to use the voice

VLAN for all sites.

3. Configure phones on HQ site to have IP addresses

120-130. Use IOS DHCP on the HQ router to accomplish

this task.

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4. Configure BR1 phones to have IP addresses of 130–140

using the IOS DHCP on the BR1 Router.

5. Configure BR2 phones to have IP addresses 210–220

using the IOS DHCP on the BR2 router.

6. Configure HQ router as an NTP master clock with

stratum 2 for local time zone in Los Angeles and

configure BR1 and BR2 to synchronize their clocks to

HQ. BR1 is in New York time zone and BR2 is in India.

CallManager and CallManager Express:

1. Register all phones according to the dial plan table

using SCCP loads. Make sure the phones will display

calling names in every scenario presented (internal

and PSTN):

2. Configure directory number 3003 as a second line on

both phones in BR2. When a call comes into 3003 it

should ring on both phones and be answered by the

first user to respond.

3. Make sure you that when the initiator of a conference

call within the BR2 site drops that the conference

call drops.

4. Make sure all phones display the same general

information aside from the line specific

configuration; this refers to HQ and BR1.

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5. Allow BR2 phone 1 to pickup calls ringing on BR2

phone 2 and vice versa. This should work for internal

and external calls.

6. Both phone 1 and phone 2 are members of VIP sales

group and are required to take calls dialed to

442076303111. The calls are expected to ring the

phones in sequential order. Both phones should ring

for 5 seconds and if the call is not answered it

should be forwarded to DN 3000.

7. Set BR2 phone 1 for extension mobility with an auto-

logout enabled. User BR2 Phone 1 should be able to

login to BR2 phone 1 and have all their settings

enabled.

8. Enable both phones at site BR2 for VoiceView Express.

9. Configure BR2 phone 2 so that when the phone is

called it rings our to 6305000 after 5 seconds and

then forwards to voicemail after 10 seconds.

Voice Gateways and Signaling:

1. Configure the HQ router as a IOS MGCP gateway using T1

PRI and NI as the line parameters.

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2. Configure BR1 router as a IOS H.323 gateway using T1

PRI and NI as the line parameters.

3. Configure BR2 as E1 PRI and make sure all BR2 numbers

are reachable.

4. Configure HQ router as a gatekeeper with the following

details:

Local zone name: voiceie

Domain name: ccbootcamp.com

Register CallManager and CME to the gatekeeper.

Register the CallManager with a tech-prefix of “1#”

and a zone called “CM”. Register CME with a tech-

prefix of “1#” and to a zone called “CME”.

Call Routing:

1. In every site, phone 1 should be allowed to dial

international and below numbers, phone 2 should only

be allowed to call long distance and below.

2. Configure the following dialing options for users in

HQ:

7+4 digits Calls to BR2

911 and 9911 Emergency

9+7 digits

The first is in the range of

2 to 9

Local

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9+1+10 digits

The first and the forth are

in the range of 2 to 9.

Long Distance

9+011+any number of digits

Please allow users to dial

the hash sign at the end of

the number in order to have

faster response

International

• Local calls from HQ should use local HQ gateway

and BR1 gateway as backup.

• Long distance calls from HQ should use local HQ

gateway as first choice and then BR1 gateway.

• Calls to BR1 PSTN numbers should be routed out

BR1 gateway (Toll Bypass), with fallback to local

HQ gateway.

• International calls to BR2 numbers, using the 7+4

digits pattern, should be routed via the

gatekeeper as VoIP and use the local HQ gateway

as backup.

• International calls to BR2 local numbers should

be routed via the gatekeeper and then via the

local HQ gateway.

• All other International calls should route out

the HQ gateway.

3. Configure the following dialing options for users in

BR1:

7+4 digits Calls to BR2

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911 and 9911 Emergency

9+7 digits

The first is in the range of 2

to 9

Local

9+1+10 digits

The first and the forth are in

the range of 2 to 9.

Long Distance

9+011+any number of digits

Please allow users to dial the

hash sign at the end of the

number in order to have faster

response

International

• Local calls from BR1 should use the local BR1 IOS

gateway and then the HQ gateway as a backup.

• Long distance calls from BR1 should use the BR1

gateway as a first choice and then the HQ gateway

as a backup.

• Calls to HQ PSTN numbers (Area Code+Prefix)

should be routed out the HQ gateway (Toll Bypass)

with fallback to the local BR1 gateway.

• International calls to BR2 numbers, using the 7+4

digits pattern, should be routed via the

gatekeeper as VoIP and use the BR1 gateway as

backup.

• International calls to BR2 local numbers should

be routed via the gatekeeper and then via the

local HQ gateway.

• All other International calls should route out

the BR1 gateway.

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4. Configure the following dialing options for users in

BR2:

7+4 digits Calls to HQ and BR1

999 Emergency

9+7 digits Local

9+0+8 digits Long Distance

9+00+any number of digits

Please allow users to dial

the hash sign at the end of

the number in order to have

faster response

International

• International calls to HQ and BR1 should use the

gatekeeper first and the local PRI as a backup.

Codec and Call Admission Control:

1. All calls within a site should use the G.711 codec.

2. All calls between sites should use the G.729 codec.

3. Allow two concurrent calls to and from BR1.

4. Allow three concurrent calls from HQ and BR1 to BR2.

High Availability Features:

1. Configure BR1 as an SRST router.

2. Make sure you only allow BR1 phone 2 to register via

SRST during a failure.

3. Preserve all PSTN dialing with using 9 as the first

digit in the string.

4. When in SRST mode, the BR1 phone 2 is not allowed to

dial international calls.

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5. Allow HQ and BR1 phones to call each other in times

when more then 2 concurrent calls are in process

between the sites.

Media Resources Management:

1. Configure conferencing for the HQ site to use hardware

based resources in the HQ gateway first then hardware

resources in the BR1 gateway.

2. Configure conferencing for the BR1 site to use

hardware based resources in the BR1 gateway first then

hardware resources in the HQ gateway.

3. Configure transcoding for the HQ site to use hardware

based resources in the HQ gateway first then hardware

resources in the BR1 gateway.

4. Configure transcoding for the BR1 site to use hardware

based resources in the BR1 gateway first then hardware

resources in the HQ gateway.

5. Configure MOH for the HQ site based on unicast and

streamed from the publisher CallManager.

6. Configure multicast MOH for BR1, based on G729 codec

from the subscriber, and streamed from the BR1

gateway.

7. Allow 5 simultaneous meet me conferences for the HQ

users, in the number range 1201-1205. Every meet me

conference should allow a maximum of ten participants.

8. Make sure that when the initiator of an Ad-Hoc

conference hangs up, the conference is allowed to

continue as long as an internal user is still

connected.

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QoS Features:

13.Configure the frame relay link between HQ and BR1 as

if it is 384k in bandwidth, make sure you enable MLP

LFI on it.

• VoIP signaling should have 5% of the line as

guarantied bandwidth.

• VoIP barrier should have 33% of the line as priority

bandwidth.

• All other traffic should be weighted fair queued.

14.Configure the frame relay link between HQ and BR2 as

if it is 1200k in bandwidth.

• VoIP signaling should have 60k of the line reserved

bandwidth.

• VoIP barrier should have 360k of the line priority

bandwidth.

• All other traffic should be weighted fair queued.

15.The BR1 gateway should send the signaling and RTP

traffic marked as CS3 and EF.

Voicemail Integrations:

1. Configure voicemail for HQ and BR1 based on the Unity

Connections server using the following parameters:

• Pilot 1170, port 1171-1174, MWI on 1180, MWI off

1181.

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2. Make sure you prevent Unity Connections from calling

to itself and creating call loops which consume the

available ports.

3. Configure a voicemail box for phones 1 and 2 in HQ and

BR1 with a phone password of 147369 for all users.

4. Configure an auto attendant for HQ with DN 1000. It

should have a customized greeting that allows dialing

an extension and walking through the users directory

by pressing 1. When users press 0 they should be

forwarded to extension 1001. During closed times,

assume normal weekday schedule, it should take a

message and store it in the HQ phone 1 mailbox.

5. Create a second line on BR1 phone 2 with extension

*2002. A call coming to the primary line should roll

to this line if the primary is busy. The user should

be able to use this line for voicemail access. Do not

use any alternate extensions.

6. Configure BR2 CME to use the local Unity Express as a

voicemail, using the following parameters:

• Pilot 3180, port 3191-3194, MWI on 3190, MWI off

3191.

7. Create voicemail for BR2 phone 1 and 2. Use a blank

phone password for all users.

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8. Create an auto attendant for BR2 that will work when

DN 3000 is dialed.

9. Make sure that calls from HQ can route via gatekeeper

and hit the auto attendant in BR2.

IPCC Express Voice Application:

1. Configure the IPCC on the Contact Center server to

work with the CallManager using the following

parameters (all passwords are “cisco” and all pin’s

are “12345”):

Jtapi username: jtapi

Resource manager user: rmjtapi

CTI ports: 1015 – 1019.

2. Configure the auto attendant script to answer when a

call arrives at 1020.

3. Configure an ICD queue that will be triggered when

dialing extension 1021. HQ phone 1 should be used as

an IP phone agent for user “HQ1” with the password

“cisco”. BR1 phone 1 should be used as a IP phone

agent with user “BR1” with a password “cisco”.

4. Callers should hear music on hold while delayed in

queue.

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5. Calls should be routed via the queue based on a skill

named “Sales”. HQ phone 1 should be skilled higher

than BR1 phone 1.

6. Before a call is routed to the queue a check should be

performed to see if any agents are logged into the

queue. If no agents are logged in then the call should

be routed to HQ phone 2.

7. If a call is received from 7029465000 then that caller

should be assigned a priority of 10 and should be

delivered to an agent before any other call.

CallManager Voice Application:

1. Create extension mobility for user “john” with a

password of “cisco”. Which will allow him to use

extension 1007 on HQ and BR1 phone 2 in each site.

Make sure that the extension mobility service is only

available on phone 2 in HQ and BR1.

Presence:

1. Integrate the Presence server with CallManager. Make

sure the HQ Phone 1 user can login via CUPC and

control the desktop phone. HQ Phone 1 user should also

be allowed to switch to softphone mode.

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2. Make sure HQ Phone 1 can see the voicemail in their

voicemail box through the CUPC client.

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Solutions:

Basic Campus Design

1. Start with “show vlan” and make sure you have all

VLANS defined in all devices. If not create them and

then add the IP interfaces to each device with the

correct IP subnet from the table provided:

Time Saver - In order to save time you should go ahead and configure the

layer 2 QoS in the HQ switch.

HQ:

HQ-3750#

HQ-3750#config t

Enter configuration commands, one per line. End with CNTL/Z.

HQ-3750(config)#vlan 20!! !

HQ-3750(config-vlan)#name Data

HQ-3750(config-vlan)#exit

HQ-3750(config)#vlan 30!! !

HQ-3750(config-vlan)#name Voice

HQ-3750(config-vlan)#exit

HQ-3750(config)#

HQ-3750(config)#interface range FastEthernet 1/0/6 - 7

HQ-3750(config-if-range)#switchport voice vlan 30

HQ-3750(config-if-range)#switchport access vlan 20

HQ-3750(config-if-range)#switchport trunk encapsulation dot1q

HQ-3750(config-if-range)#switchport mode trunk

HQ-3750(config-if-range)#auto qos voip cisco-phone

HQ-3750(config)#interface range FastEthernet 1/0/1 - 5

HQ-3750(config-if-range)#switchport host

HQ-3750(config-if-range)#auto qos voip trust

HQ-3750(config)#interface FastEthernet 1/0/24

HQ-3750(config-if)#auto qos voip trust

HQ-3750(config)#interface FastEthernet0/0.20

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HQ-3750(config-if)#encapsulation dot1Q 20

HQ-3750(config-if)#ip address 10.1.20.1 255.255.255.0

HQ-3750(config)#exit

HQ-3750(config)#interface FastEthernet0/0.30

HQ-3750(config-if)#encapsulation dot1Q 30

HQ-3750(config-if)#ip address 10.1.30.1 255.255.255.0

! Tip - Don’t forget to add the new networks into the routing protocol.

HQ-3750(config)#router ospf 1

HQ-3750(config-router)#network 10.1.20.0 0.0.0.255 area 0

HQ-3750(config-router)#network 10.1.30.0 0.0.0.255 area 0

BR1:

BR1#vlan database

BR1(vlan)#vlan 120 name Data state active

VLAN 120 added:

Name: Data

State ACTIVE

BR1(vlan)#vlan 130 name Voice state active

VLAN 130 added:

Name: Voice

State ACTIVE

BR1(vlan)#exit

APPLY completed.

Exiting....

! Tip - The “vlan database” command has been deprecated so you can now use

the “vlan” command under global configuration mode as well.

BR1#config t

Enter configuration commands, one per line. End with CNTL/Z.

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BR1(config)#int vlan 120

BR1(config-if)#ip address 10.1.120.1 255.255.255.0

BR1(config-if)#description Data

BR1(config-if)#int vlan 130

BR1(config-if)#ip address 10.1.130.1 255.255.255.0

BR1(config-if)#description Voice

BR1(config-if)#exit

BR1(config)#interface range FastEthernet 0/3/0 - 1

BR1(config-if-range)#switchport voice vlan 130

BR1(config-if-range)#switchport access vlan 120

BR1(config-if-range)#no shut

! Tip - Don’t forget to add the new networks into the routing protocol.

BR1(config)#router ospf 1

BR1(config-router)#network 10.1.120.0 0.0.0.255 area 0

BR1(config-router)#network 10.1.130.0 0.0.0.255 area 0

BR2:

BR2#vlan database

BR2(vlan)#vlan 220 name Data state active

VLAN 220 added:

Name: Data

State ACTIVE

BR2(vlan)#vlan 230 name Voice state active

VLAN 230 added:

Name: Voice

State ACTIVE

BR2(vlan)#exit

APPLY completed.

Exiting....

BR2#config t

Enter configuration commands, one per line. End with CNTL/Z.

BR2(config)#int vlan 220

BR2(config-if)#ip address 10.1.220.1 255.255.255.0

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BR2(config-if)#description Data

BR2(config-if)#int vlan 230

BR2(config-if)#ip address 10.1.230.1 255.255.255.0

BR2(config-if)#description Voice

BR2(config-if)#exit

BR2(config)#interface range FastEthernet 0/1/0 - 1

BR2(config-if-range)#switchport voice vlan 230

BR2(config-if-range)#switchport access vlan 220

BR2(config-if-range)#no shut

! Tip - Don’t forget to add the new networks into the routing protocol.

BR2(config)#router ospf 1

BR2(config-router)#network 10.1.220.0 0.0.0.255 area 0

BR2(config-router)#network 10.1.230.0 0.0.0.255 area 0

2. Validate your configurations with the “show

vlan” ( or “show current” under the vlan database)

and “show cdp neighbors” commands.

!

HQ-3750#show vlan

VLAN Name Status Ports

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

1 default active ! Fa1/0/1, Fa1/0/2, Fa1/0/3, Fa1/0/4,

! ! ! ! ! ! Fa1/0/5, Fa1/0/8, Fa1/0/9, Fa1/0/10,

! ! ! ! ! ! Fa1/0/11, Fa1/0/12, Fa1/0/13, Fa1/0/14,

! ! ! ! ! ! Fa1/0/15, Fa1/0/16, Fa1/0/17, Fa1/0/18,

! ! ! ! ! ! Fa1/0/19, Fa1/0/20, Fa1/0/21, Fa1/0/22,

! ! ! ! ! ! Fa1/0/23, Gi1/0/1, Gi1/0/2

20 Data active

30 Voice active

1002 fddi-default act/unsup

1003 token-ring-default act/unsup

1004 fddinet-default act/unsup

1005 trnet-default act/unsup

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HQ-3750#show cdp neigh

Capability Codes: R - Router, T - Trans Bridge, B - Source Route Bridge

S - Switch, H - Host, I - IGMP, r - Repeater, P - Phone

Device ID Local Intrfce Holdtme Capability Platform Port ID

CCMSUB Fas 1/0/1 152 H none foundeth0

CCMPUB Fas 1/0/1 157 H none foundeth0

HQ_Rtr Fas 1/0/24 140 R S I 2811 Fas 0/0.1

uccx. Fas 1/0/5 172 H Win2000 SeEth 1/1

presence Fas 1/0/4 145 H none foundeth0

SEP003094C27864 Fas 1/0/6 138 H P IP Phone 7Port 1

SEP003094C2DA38 Fas 1/0/7 132 H P IP Phone 7Port 1

SEP0002FDAEFDDF Fas 1/0/8 159 H P IP Phone 7Port 1

Unity Fas 1/0/3 162 H none foundeth0

PSTN-WAN Fas 1/0/22 124 R S I 2811 Fas 0/0

HQ-3750#

!Note - The publisher and subscriber are on the same physical box so they will

show as if they are connected to the same port on the switch. This is normal

and you should configure FastEthernet 1/0/2 as if the subscriber is plugged

into that port.

BR1#vlan database

BR1(vlan)#show current

VLAN ISL Id: 1

Name: default

Media Type: Ethernet

VLAN 802.10 Id: 100001

State: Operational

MTU: 1500

Translational Bridged VLAN: 1002

Translational Bridged VLAN: 1003

VLAN ISL Id: 120

Name: Data

Media Type: Ethernet

VLAN 802.10 Id: 100220

State: Operational

MTU: 1500

VLAN ISL Id: 130

Name: Voice

Media Type: Ethernet

VLAN 802.10 Id: 100230

State: Operational

MTU: 1500

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BR1#show cdp neigh

Capability Codes: R - Router, T - Trans Bridge, B - Source Route Bridge

S - Switch, H - Host, I - IGMP, r - Repeater

Device ID Local Intrfce Holdtme Capability Platform Port ID

HQ_Rtr Ser 0/1/0.1 161 R S I 2811 Ser 0/2/0.1

SEP0016C72D3DCC Fas 0/3/1 146 H IP Phone Port 1

SEP003094C3DCF9 Fas 0/3/0 143 H IP Phone Port 1

BR1#

BR2# show cdp neigh

Capability Codes: R - Router, T - Trans Bridge, B - Source Route Bridge

S - Switch, H - Host, I - IGMP, r - Repeater

Device ID Local Intrfce Holdtme Capability Platform Port ID

HQ_Rtr Ser 0/2/0.1 133 R S I 2811 Ser 0/2/0.2

SEP001F6C7FD6C1 Fas 0/1/1 167 H IP Phone Port 1

SEP001F6C7F722B Fas 0/1/0 167 H IP Phone Port 1

BR2#

BR2#vlan database

BR2(vlan)#show current

VLAN ISL Id: 1

Name: default

Media Type: Ethernet

VLAN 802.10 Id: 100001

State: Operational

MTU: 1500

Translational Bridged VLAN: 1002

Translational Bridged VLAN: 1003

VLAN ISL Id: 220

Name: Data

Media Type: Ethernet

VLAN 802.10 Id: 100220

State: Operational

MTU: 1500

VLAN ISL Id: 230

Name: Voice

Media Type: Ethernet

VLAN 802.10 Id: 100230

State: Operational

MTU: 1500

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3. In order to accomplish this we need to first create

the dhcp excluded address and then define the DHCP

pool:

HQ_Rtr(config)#ip dhcp excluded-address 10.1.30.1 10.1.30.119

HQ_Rtr(config)#ip dhcp excluded-address 10.1.30.131 10.1.30.255

HQ_Rtr(config)#ip dhcp pool IPPHONE

HQ_Rtr(config-config)#network 10.1.30.0 255.255.255.0

HQ_Rtr(config-config)#default-router 10.1.30.1

HQ_Rtr(config-config)#option 150 ip 10.1.200.21 10.1.200.25

Time Saver - In order to save time you should use notepad to copy and

paste configurations which may be the same or need little modification.

4. We can use the same configuration from the last task

but modify for BR1:

BR1(config)#ip dhcp excluded-address 10.1.130.1 10.1.130.129

BR1(config)#ip dhcp excluded-address 10.1.130.141 10.1.130.255

BR1(config)#ip dhcp pool IPPHONE

BR1(config-config)#network 10.1.130.0 255.255.255.0

BR1(config-config)#default-router 10.1.130.1

BR1(config-config)#option 150 ip 10.1.200.21 10.1.200.25

5. Perform the same for BR2:

BR2(config)#ip dhcp excluded-address 10.1.230.1 10.1.230.209

BR2(config)#ip dhcp excluded-address 10.1.230.221 10.1.230.255

BR2(config)#ip dhcp pool IPPHONE

BR2(config-config)#network 10.1.230.0 255.255.255.0

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BR2(config-config)#default-router 10.1.230.1

BR2(config-config)#option 150 ip 10.1.230.1

6. Next configure each router for the appropriate time

zone and NTP settings:

HQ_Rtr(config)#clock timezone PST -8

HQ_Rtr(config)#clock summer-time PDT recurring

HQ_Rtr(config)#ntp master 2

BR1(config)#clock timezone EST -5

BR1(config)#clock summer-time EDT recurring

BR1(config)#ntp server 10.10.32.1

BR2(config)#clock timezone UTC +5 30

BR2(config)#clock summer-time PDT recurring

BR2(config)#ntp server 10.10.32.1

! Tip - Use can use your candidate PC to look up the time zone of most

countries but most likely they will specify for you.

! Tip - Don’t forget to set the time zone settings in CallManager as it may not be

mentioned in the phone setup tasks.

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CallManager and CallManager Express:

1. This tasks requires a lot of configuration as this is

the base line for everything you do in CallManager. We

will setup everything from device pools, locations,

regions, media resource lists and groups, route lists

and route groups, partitions and calling search

spaces.

!

Lets start off with service activation first as

nothing will work if we don’t have services activated:

Make sure you enable services on both servers and all

services except the “Cisco Messaging Interface”.

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!Tip - Enabling all services as stated above will ensure that all needed services

are activated. Unless you are specifically told to only enable needed services

then this strategy will allow you to make sure all the required services are

running.

Next we need to start configuring the required

CallManager settings needed to allow for the

registration of HQ and BR1 phones.

Make sure the proper servers are defined:

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Next create a CallManager group that will allow for

devices to register to the subscriber and then the

publisher:

When finished you should see the following to groups:

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! Note - The requirement for registration order is normally on the first page of

instructions.

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Next we need to create the regions:

We need to make sure the settings between the regions

are set appropriately:

Time Saver - This is where reading through the whole lab will save a lot of

time as we can set the proper region settings and never return to this page.

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Page 775: Ccie Voice Advanced Lab Workbook

Now we need to create the HQ and BR1 Date/Time groups:

! Note - This is also where we could set the date/time format if asked during a

specific task.

Next we need to create the locations:

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Time Saver - This is where reading through the whole lab will save a lot of

time as we can set the proper location settings and never return to this page.

We now need to create the media resource groups and

lists:

!Tip - In reading the lab we know that there is a need to separate the MOH

servers and it is always a good idea to separate hardware from software

resources.

We can then create the needed media resource lists:

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Page 779: Ccie Voice Advanced Lab Workbook

Below is the full list of the media resource lists:

We now need to create the device pools:

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Time Saver - We know from reading the lab that one phone will not be

allowed to register in SRST. We can create the device pool for this function

now in order to save time later.

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Here is the final list of device pools:

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Next we need to check the Enterprise Parameters and

make changes as needed:

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!Tip - Changing the host name to an IP address in the URL settings will prevent

authentication and IP phone service failures. This is highly recommended

whether you have a DNS server or not!

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Page 785: Ccie Voice Advanced Lab Workbook

We can also create the AAR groups now as this will

help to cut down the number of times we need to visit

the phone configuration page:

Once we have both groups created we need to set the

dial prefix according to the dial plan:

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Now we will need to create the the partitions and

calling search spaces that will be needed for the lab:

On the next page is a table of calling search spaces

and the partitions that belong to each calling search

space.

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!"##$%&'()"*+,'(-"+)'

."/)' 0)1+*$-2$3%' 4352)'6"*2$2$3%1'73*'2,$1'!"##$%&'()"*+,'(-"+)'

!""#$%&#''%( !""#$%&#''%( )*+$%&+''%(

!""#$%&#,-*( !""#$%&#,-*( )*+$%&+.&&(

)*+$%&+,/012/34(

)*+$%&+*566$7)'""(

)*+89+,/012/34(

)*+$%:+,/012/34(

)*+$%&+6;<34(

)*+$%&+6=(

)*+$%&+,-*(

!""#$%&#,-*>%-'6( !""#$%&#,-*>%-'6( )*+$%&+.&&(

)*+$%&+,/012/34(

)*+$%&+*566$7)'""(

)*+89+,/012/34(

)*+$%:+,/012/34(

!""#$%&#6=( !""#$%&#6=( )*+$%&+.&&(

)*+$%&+,/012/34(

)*+$%&+*566$7)'""(

)*+89+,/012/34(

)*+$%:+,/012/34(

)*+$%&+6;<34(

)*+$%&+6=(

!""#$%&#65!'6( !""#$%&#65!'6( )*+$%&+.&&(

)*+$%&+,/012/34(

)*+$%&+*566$7)'""(

)*+89+,/012/34(

)*+$%:+,/012/34(

)*+$%&+6;<34(

!""#$%:( !""#$%:( )*+$%&+,/012/34(

)*+$%:+,/012/34(

)*+89+,/012/34(

)*+$%:+*566$7)'""(

!""#89#''%( !""#89#''%( )*+89+''%(

!""#89#,-*( !""#89#,-*( )*+89+.&&(

)*+89+,/012/34(

)*+89+*566$7)'""(

)*+$%&+,/012/34(

)*+$%:+,/012/34(

)*+89+6;<34(

)*+89+6=(

)*+89+,-*(

!""#89#,-*>%-'6( !""#89#,-*>%-'6( )*+89+.&&(

)*+89+,/012/34(

)*+89+*566$7)'""(

)*+$%&+,/012/34(

)*+$%:+,/012/34(

!""#89#6=( !""#89#6=( )*+89+.&&(

)*+89+,/012/34(

)*+89+*566$7)'""(

)*+$%&+,/012/34(

)*+$%:+,/012/34(

)*+89+6;<34(

)*+89+6=(

!""#89#65!'6( !""#89#65!'6( )*+89+.&&(

)*+89+,/012/34(

)*+89+*566$7)'""(

)*+$%&+,/012/34(

)*+$%:+,/012/34(

)*+89+6;<34(

!""#?5,!>@',6( !""#?5,!>@',6( )*+$%&+,/012/34(

)*+89+,/012/34(

(

(

We should now be able to add the phones per the

requirements in the table.

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HQ Phone 1:

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HQ Phone 2:

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BR1 Phone 1:

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BR1 Phone 2:

!Note - In reading the lab we know the BR1 Phone 2 requires another extension

“*2002”. We can create that as part of the initial phone setup which we have

done here or you could wait till that specific task.

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BR1 Phone 2 (Line 2):

! Tip - Most of the phone settings per phone are the same per site except for the

calling search space.

Time Saver - In order to save time you can use auto registration on the

callmanager and then modify each phone to the correct settings.

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We can now start the registration of phones within the

BR2 site. We will first need to define all the

required settings underneath the “telephony-service”

configuration. We can then create the phones and their

associated DN’s:

BR2(config)#telephony-service

BR2(config-telephony)#authentication credential admin cisco

BR2(config-telephony)#max-ephones 2

BR2(config-telephony)#max-dn 7

BR2(config-telephony)#system message Your current options

BR2(config-telephony)#ip source-address 10.1.230.1 port 2000

BR2(config-telephony)#load 7941 term41.default.loads

BR2(config-telephony)#voicemail 3180

BR2(config-telephony)#max-conferences 8 gain -6

BR2(config-telephony)#call-forward pattern .T

BR2(config-telephony)#dn-webedit

BR2(config-telephony)#time-webedit

BR2(config-telephony)#transfer-system full-blind

BR2(config-telephony)#transfer-pattern 9.T

BR2(config-telephony)#secondary-dialtone 9

BR2(config-telephony)#create cnf-files

BR2(config-telephony)#exit

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Now we can configure the phones and their DN’s:

BR2(config)#ephone-dn 1

BR2(config-ephone-dn)#number 3001

BR2(config-ephone-dn)#pickup-group 1

BR2(config-ephone-dn)#description 6303001

BR2(config-ephone-dn)#name BR2 Phone 1

BR2(config-ephone-dn)#call-forward noan 3180 timeout 5

BR2(config)#ephone-dn 2

BR2(config-ephone-dn)#number 3002

BR2(config-ephone-dn)#pickup-group 1

BR2(config-ephone-dn)#description 6303002

BR2(config-ephone-dn)#name BR2 Phone 2

We can also configure the extra DN’s specified for the

shared line on BR2 phone 1 and 2:

BR2(config)#ephone-dn 6 dual-line

BR2(config-ephone-dn)# number 3003

BR2(config-ephone-dn)# pickup-group 1

BR2(config-ephone-dn)# description 6303003

BR2(config)#ephone-dn 7 dual-line

BR2(config-ephone-dn)# number 3004

BR2(config-ephone-dn)# pickup-group 1

We can now configure the phones knowing that phone 1

will be using extension mobility:

BR2(config)#ephone 1

BR2(config-ephone)#conference drop-mode creator

BR2(config-ephone)#mac-address 001F.6C7F.722B

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BR2(config-ephone)#type 7941GE

BR2(config-ephone)#logout-profile 1

BR2(config)#ephone 2

BR2(config-ephone)#conference drop-mode creator

BR2(config-ephone)#mac-address 001F.6C7F.D6C1

BR2(config-ephone)#username "BR2Phone2"

BR2(config-ephone)#type 7941GE

BR2(config-ephone)#button 1:2 2o6,7

Next we need to allow the SCCP firmware to be

downloaded from the router flash:

BR2(config)#tftp-server flash:apps41.8-3-2-27.sbn BR2(config)#tftp-server flash:cnu41.8-3-2-27.sbn BR2(config)#tftp-server flash:cvm41sccp.8-3-2-27.sbn BR2(config)#tftp-server flash:dsp41.8-3-2-27.sbn BR2(config)#tftp-server flash:jar41sccp.8-3-2-27.sbn

!Tip - If you like to use the CME and CUE web interface the the following

commands should be entered at this stage to prepare you to use the web

interface.

BR2(config)#ip http server

BR2(config)#no ip http secure-server

BR2(config)#ip http path flash:

BR2(config)#telephony-service

BR2(config-telephony)#web admin system name admin password cisco

BR2(config-telephony)#url services http://10.1.230.2/voiceview/common/

login.do (Inserted by default in most cases)

BR2(config-telephony)#url authentication http://10.1.230.1/CCMCIP/

authenticate.asp (Inserted by default in most cases)

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2. This was already completed in the above solution with the

ephone-dn 6 and 7 configuration and the addition of this

button with overlay to phone 2 ephone. Adding this to

phone 1 will come later during the extension mobility

task.

3. This was already completed in the above solution with the

“conference drop-mode creator” command under the ephone

configuration for phone 2. Adding this to phone 1 will

come later during the extension mobility task.

4. This was accomplished in task 1 by setting the alerting

name, caller id, external phone number mask, and date/

time group format the same in CallManager.

5. This was accomplished in task 1 by setting setting the

“pickup-group 1” command under all the ephone-dn’s.

6. This is accomplished with the following configuration:

BR2(config)#ephone-hunt 1 sequential

BR2(config-ephone-hunt)#pilot 3111

BR2(config-ephone-hunt)#list 3001, 3003

BR2(config-ephone-hunt)#final 3000

BR2(config-ephone-hunt)#timeout 5, 5

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7. This is accomplished through creating a logout profile

and a user profile for phone 1:

BR2(config)#voice logout-profile 1

BR2(config-voice-profile)#pin 12345

BR2(config-voice-profile)#user br2phone3 password cisco

BR2(config-voice-profile)#number 3003,3004 type overlay

BR2(config)#voice user-profile 1

BR2(config-voice-profile)#pin 12345

BR2(config-voice-profile)#user BR2Phone1 password cisco

BR2(config-voice-profile)#number 3001 type normal

BR2(config-voice-profile)#number 3003,3004 type overlay

8. In this version of CUE and CME voice view is enabled by

default. One step that needs to take place is adding the

CME authentication URL as the backup URL for the voice

view service in CUE:

CUE(config)#service phone-authentication CUE(config-service)#fallback-url http://10.1.230.1/CCMCIP/

authenticate.asp CUE(config-service)#end phone-authentication

9. The is simple to accomplish by adding the following under

the ephone-dn:

BR2(config)#ephone-dn 2

BR2(config-ephone-dn)#mobility

BR2(config-ephone-dn)#snr 96305000 delay 5 timeout 10 cfwd-noan 3180

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!Note - Though the task asks us to allow the number to be reach by the full

E164 number we can perform this action later when setting up the call routing

as all calls will need to be striped to four digits.

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Page 811: Ccie Voice Advanced Lab Workbook

Voice Gateways and Signaling

1. In order to enable the HQ router as an MGCP gateway we

need the following configuration on the HQ router:

HQ_Rtr(config)#network-clock-participate wic 0

HQ_Rtr(config)#isdn switch-type primary-ni

HQ_Rtr(config)#voice-card 0

HQ_Rtr(config-voicecard)#dspfarm

HQ_Rtr(config-voicecard)#dsp services dspfarm

HQ_Rtr(config-voicecard)#exit

HQ_Rtr(config)#controller T1 0/0/0

HQ_Rtr(config-controller)#cablelength long 0db

HQ_Rtr(config-controller)#pri-group timeslots 1-3,24 service mgcp

HQ_Rtr(config-controller)#exit

HQ_Rtr(config)#interface Serial0/0/0:23

HQ_Rtr(config-if)# isdn bind-l3 ccm-manager

HQ_Rtr(config-if)#exit

HQ_Rtr(config)#ccm-manager redundant-host 10.1.200.21

HQ_Rtr(config)#ccm-manager mgcp

HQ_Rtr(config)#mgcp

HQ_Rtr(config)#mgcp call-agent 10.1.200.25 service-type mgcp version 0.1

HQ_Rtr(config)#mgcp fax t38 ecm

HQ_Rtr(config)#mgcp

! Note - In IOS release 12.4 the dial-peer no longer needs to be created with the

service command.

Time Saver - If you have multiple MGCP or H.323 gateways then use

notepad to copy and paste common configuration components.

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Next we will need to configure the gateway in

CallManager:

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Page 815: Ccie Voice Advanced Lab Workbook

!Tip - If you have set or been asked to set a domain name on the router the

device name for MGCP would be the router name plus the domain name. So in

our case it could be “HQ_Rtr.voiceie.com” if we had “voiceie.com” as a

domain name on the router.

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Page 816: Ccie Voice Advanced Lab Workbook

2. In order to enable the BR1 router as an H.323 gateway we

need the following configuration on the BR1 router:

BR1(config)#network-clock-participate wic 1

BR1(config)#isdn switch-type primary-ni

BR1(config)#voice-card 0

BR1(config-voicecard)#dsp services dspfarm

BR1(config-voicecard)#exit

BR1(config)#voice translation-rule 1

BR1(cfg-translation-rule)#rule 1 /\(^702333\)\(2...\)/ /\2/

BR1(cfg-translation-rule)#exit

BR1(config)#voice translation-profile STRIP

BR1(cfg-translation-profile)#translate called 1

BR1(cfg-translation-profile)#exit

BR1(config)#voice-port 0/1/0:23

BR1(config-voiceport)#translation-profile incoming STRIP

BR1(config-voiceport)#exit

BR1(config)#controller T1 0/1/0

BR1(config-controller)#framing esf

BR1(config-controller)# linecode b8zs

BR1(config-controller)# pri-group timeslots 1-3,24

BR1(config-controller)# exit

BR1(config)#interface Loopback0

BR1(config-if)#h323-gateway voip interface

BR1(config-if)#h323-gateway voip bind srcaddr 10.10.32.2

BR1(config-if)#exit

BR1(config)#interface Serial0/1/0:23

BR1(config-if)#isdn outgoing display-ie

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Next we will need to setup BR1 as an H.323 gateway in

CallManager:

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!Tip - The most common issue with H.323 gateways is that IP address specified

in CallManager is not the IP address that the H.323 process in the router is

binding. Make sure these IP address match!

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Page 819: Ccie Voice Advanced Lab Workbook

3. In order to enable the BR2 router as an H.323 gateway we

need the following configuration on the BR2 router:

BR2(config)#network-clock-participate wic 0

BR2(config)#isdn switch-type primary-net5

BR2(config)#voice translation-rule 1

BR2(cfg-translation-rule)#rule 1 /\(^44207630\)\(3...\)/ /\2/

BR2(cfg-translation-rule)#exit

BR2(config)#voice translation-profile STRIP

BR2(cfg-translation-profile)#translate called 1

BR2(cfg-translation-profile)#exit

BR2(config)#voice-card 0

BR2(config-voicecard)#dspfarm

BR2(config-voicecard)#dsp services dspfarm

BR2(config-voicecard)#exit

BR2(config)#controller E1 0/0/0

BR2(config-controller)#pri-group timeslots 1-3,16

BR2(config-controller)#exit

BR2(config)#interface Serial0/0/0:15

BR2(config-if)#isdn outgoing display-ie

BR2(config-if)#exit

BR2(config)#voice-port 0/0/0:15

BR2(config-voiceport)#translation-profile incoming STRIP

4. In order to setup the gatekeeper we need to perform the

following configuration on the HQ router:

HQ_Rtr(config)#gatekeeper

HQ_Rtr(config-gk)#zone local CM ccbootcamp.com 10.10.32.1

HQ_Rtr(config-gk)#zone local CME ccbootcamp.com

HQ_Rtr(config-gk)#gw-type-prefix 1#* default-technology

HQ_Rtr(config-gk)#no shutdown

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Next we need to enable CallManager to register to the

gatekeeper through a H.225 gatekeeper controlled

trunk. We can accomplish this by adding the gatekeeper

and then the trunk in CallManager:

!Tip - If you would like to lower the time it takes for gatekeeper to register

after a loss of connectivity then change the “Registration Retry Timeout” to 30

seconds.

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Page 821: Ccie Voice Advanced Lab Workbook

Now we need to add the trunk:

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! Tip - You can validate registration with the “show gatekeeper endpoints”

command on the HQ router.

HQ_Rtr#show gatekeeper endpoints

GATEKEEPER ENDPOINT REGISTRATION

================================

CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags

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

10.1.200.21 33072 10.1.200.21 32794 CM H323-GW

H323-ID: BR2_1

Voice Capacity Max.= Avail.= Current.= 0

10.1.200.25 33077 10.1.200.25 32785 CM H323-GW

H323-ID: BR2_2

Voice Capacity Max.= Avail.= Current.= 0

Total number of active registrations = 4

HQ_Rtr#

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Page 824: Ccie Voice Advanced Lab Workbook

Next we will need to configure the BR2 router is

register with the gatekeeper. This requires the

following configuration:

BR2(config)#interface Loopback0

BR2(config-if)#h323-gateway voip interface

BR2(config-if)#h323-gateway voip id CME ipaddr 10.10.32.1 1719

BR2(config-if)#h323-gateway voip h323-id CME

BR2(config-if)#h323-gateway voip tech-prefix 1#

BR2(config-if)#h323-gateway voip bind srcaddr 10.10.32.3

BR2(config-if)#exit

BR2(config)#gateway

! Tip - Once again you can verify with the “show gatekeeper endpoints”

command on the HQ router.

HQ_Rtr#show gatekeeper endpoints

GATEKEEPER ENDPOINT REGISTRATION

================================

CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags

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

10.10.32.3 1720 10.10.32.3 57205 CM H323-GW

H323-ID: CME

Voice Capacity Max.= Avail.= Current.= 0

10.1.200.21 33072 10.1.200.21 32794 CM H323-GW

H323-ID: BR2_1

Voice Capacity Max.= Avail.= Current.= 0

10.1.200.25 33077 10.1.200.25 32785 CM H323-GW

H323-ID: BR2_2

Voice Capacity Max.= Avail.= Current.= 0

Total number of active registrations = 4

HQ_Rtr#

!Note - You will notice that the directory numbers from the SIP CME did not

register with gatekeeper. This is to be expected as SIP CME phones cannot be

auto registered to gatekeeper.

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Page 826: Ccie Voice Advanced Lab Workbook

Call Routing

1. The has already been accomplished for HQ and BR1 when

we setup the phones previously by giving them the

proper calling search space. To enable class of

service for the BR2 site we need to create the

following COR list:

dial-peer cor custom

name Local

name LD

name INT

name Emergency

dial-peer cor list Emergency

member Emergency

dial-peer cor list Local

member Local

dial-peer cor list Int

member INT

dial-peer cor list LD

member LD

dial-peer cor list Phone1

member Local

member LD

member INT

member Emergency

dial-peer cor list Phone2

member Local

member LD

member Emergency

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Now that we have created the COR list we need to

assign the proper COR list to the voice register pool

of the proper phone:

BR2(config)#ephone-dn 1

BR2(config-ephone-dn)#cor incoming Phone1

BR2(config)#ephone-dn 2

BR2(config-ephone-dn)#cor incoming Phone2

2. In order to accomplish the next two tasks we need to

create all the route groups and route lists first. We

will first need to create the route groups:

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Page 828: Ccie Voice Advanced Lab Workbook

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Page 829: Ccie Voice Advanced Lab Workbook

Then next the route lists in the following tables with

the proper settings:

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Page 830: Ccie Voice Advanced Lab Workbook

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Page 831: Ccie Voice Advanced Lab Workbook

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Page 832: Ccie Voice Advanced Lab Workbook

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Page 833: Ccie Voice Advanced Lab Workbook

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Page 834: Ccie Voice Advanced Lab Workbook

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2<>>#?!:<&)B!C&<+$D*&E<)'*+$!

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Page 835: Ccie Voice Advanced Lab Workbook

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!"#$%&'()$&

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$$

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!"#$%&'()$&

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:$#!9@BB'+D!6@&)EH$!?I)#&+@B!;@$J! 3+!

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6&#F'I!"'D')$!K3=)D*'+D!9@BB$L! !

9@BB#C!6@&)E!2&@+$F*&G@)'*+$!

"'$%@&C!"'D')$! 6&#"*)!

9@BB#C!6@&)E!2&@+$F*&G!;@$J! !

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6&#F'I!"'D')$!K3=)D*'+D!9@BB$L! !

9@BB#C!6@&)E!2&@+$F*&G@)'*+$!

"'$%@&C!"'D')$! 6&#"*)!

9@BB#C!6@&)E!2&@+$F*&G!;@$J! !

6&#F'I!"'D')$!K3=)D*'+D!9@BB$L! NOOPP1NQ!

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9@BB'+D!6@&)E!2&@+$F*&G!;@$J! !

6&#F'I!"'D')$!K3=)D*'+D!9@BB$L! !

9@BB#C!6@&)E!2&@+$F*&G@)'*+$!

"'$%@&C!"'D')$! 6&#"*)!

9@BB#C!6@&)E!2&@+$F*&G!;@$J! !

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!"#$%&'()$&

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!"#$%&'(%)%*+',-.*)/%0)'1233+4' '

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'

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Page 839: Ccie Voice Advanced Lab Workbook

!"#$%&'()$&

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! "#$%&'!(&)&*+!,-.*)/&0)!1233+4! !

1233$5!"2#*6!7#20+%/#82*&/0+!

(&+92#5!(&)&*+! "#$(/*!

1233$5!"2#*6!7#20+%/#8!:2+;! !

"#$%&'!(&)&*+!,-.*)/&0)!1233+4! !

:$8<$#+! =>?=@>?(=AB>CDEFG*#!,H33!"/#*+4!

!

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@+$!1233&0)!"2#*6I+!J'*$#023!:2+;! -0!

1233&0)!"2#*6!7#20+%/#8!:2+;! !

"#$%&'!(&)&*+!,-.*)/&0)!1233+4! !

1233$5!"2#*6!7#20+%/#82*&/0+!

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"#$%&'!(&)&*+!,-.*)/&0)!1233+4! KAL>M!

:$8<$#+! A>NA>NOMNM!,H33!"/#*+4!

!!!

!

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Page 840: Ccie Voice Advanced Lab Workbook

A summary of all the route lists is below:

We can now create all the route patterns for the HQ

call routing and point them to the proper route list:

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! Note - All the digit manipulation and phone number masking is done in the

route lists and therefore should not be done in the route pattern.

Next we need to add the proper call routing to the

gatekeeper on the HQ gateway:

HQ_Rtr(config-gk)#zone prefix CME 3...

HQ_Rtr(config-gk)#zone prefix CM 7....

HQ_Rtr(config-gk)#zone prefix CME 96......

3. We can now use the previous created route groups and

route lists to setup the BR1 call routing by adding

the following route patterns:

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Page 843: Ccie Voice Advanced Lab Workbook

Now since the BR1 gateway is an H.323 gateway we need

to add the proper dial-peers to the BR1 gateway to

complete the call routing. We may also want to take

into account the SRST task for maintaining class of

service and create the COR list at the same time:

dial-peer cor custom

name Local

name LD

name INT

name Emergency

dial-peer cor list Emergency

member Emergency

dial-peer cor list Local

member Local

dial-peer cor list Int

member INT

dial-peer cor list LD

member LD

dial-peer cor list Phone1

member Local

member LD

member INT

member Emergency

dial-peer cor list Phone2

member Local

member LD

member Emergency

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Page 845: Ccie Voice Advanced Lab Workbook

We can now create the dial-peers with the COR already

configured:

dial-peer voice 1 pots

incoming called-number .

direct-inward-dial

dial-peer voice 2000 voip

preference 1

destination-pattern 2...

session target ipv4:10.1.200.25

ip qos dscp cs3 signaling

dial-peer voice 2001 voip

preference 2

destination-pattern 2...

session target ipv4:10.1.200.21

ip qos dscp cs3 signaling

dial-peer voice 7 pots

corlist outgoing Local

destination-pattern 9[2-9]......

port 0/1/0:23

forward-digits 7

dial-peer voice 11 pots

corlist outgoing LD

destination-pattern 91[2-9]..[2-9]......

port 0/1/0:23

forward-digits 11

!

dial-peer voice 911 pots

corlist outgoing Emergency

destination-pattern 911

port 0/1/0:23

forward-digits all

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Page 846: Ccie Voice Advanced Lab Workbook

!

dial-peer voice 9911 pots

corlist outgoing Emergency

destination-pattern 9911

port 0/1/0:23

forward-digits 3

!

dial-peer voice 9011 pots

corlist outgoing Int

destination-pattern 9011T

port 0/1/0:23

prefix 011

4. We can now setup the call routing for BR2 with the

following configuration on the BR2 router. We first

need to allow all the communications that maybe needed

in our environment:

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

supplementary-service h450.12

no supplementary-service sip refer

fax protocol cisco

h323

sip

registrar server expires max 1200 min 300

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Page 847: Ccie Voice Advanced Lab Workbook

Now we can create all the needed dial-peers with the

appropriate COR list which were previously defined:

dial-peer voice 1 pots

incoming called-number .

direct-inward-dial

!

dial-peer voice 2 pots

corlist outgoing Local

destination-pattern 90[1-9].......

port 0/0/0:15

prefix 0

!

dial-peer voice 1000 voip

preference 1

destination-pattern 71...

session target ras

!

dial-peer voice 1001 pots

preference 2

destination-pattern 71...

port 0/0/0:15

prefix 0017029461

!

dial-peer voice 2000 voip

preference 1

destination-pattern 72...

session target ras

!

dial-peer voice 2001 pots

preference 2

destination-pattern 72...

port 0/0/0:15

prefix 0017023332

!

dial-peer voice 7 pots

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Page 848: Ccie Voice Advanced Lab Workbook

corlist outgoing Local

destination-pattern 9[1-9]......

port 0/0/0:15

forward-digits 7

!

dial-peer voice 999 pots

corlist outgoing Emergency

destination-pattern 999

port 0/0/0:15

forward-digits all

!

dial-peer voice 900 pots

corlist outgoing Int

destination-pattern 900.T

port 0/0/0:15

prefix 00

We also need to add a translation pattern to strip the

“7” as the call comes into callmanager from BR2:

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Page 849: Ccie Voice Advanced Lab Workbook

We will also need to enable transcoding on the BR2

gateway to support calls transgressing from HQ/BR1 to

BR2:

sccp local FastEthernet0/0.230

sccp ccm 10.1.230.1 identifier 1 priority 1 version 3.1

sccp

sccp ccm group 1

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Page 850: Ccie Voice Advanced Lab Workbook

associate ccm 1 priority 1

associate profile 1 register TRANSCODE

dspfarm profile 1 transcode

codec g711ulaw

codec g711alaw

codec g729r8

codec g729abr8

codec g729ar8

codec g729br8

maximum sessions 4

associate application SCCP

telephony-service

sdspfarm units 2

sdspfarm transcode sessions 4

sdspfarm tag 1 TRANSCODE

!Tip - At this point you can start testing your calls to make sure they all

complete correctly. In order to test redundancy you can shutdown the

gatekeeper or the corresponding T1/E1 controller to test.

Time Saver - By saving the testing until your call routing is complete you

can save time by not interrupting your configuration and then fixing as you

test.

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Page 851: Ccie Voice Advanced Lab Workbook

Codec and Call Admission Control

1. This task was already accomplished earlier when we

created the regions.

2. This task was already accomplished earlier when we

created the regions.

3. This task was completed when we create the locations

earlier.

4. In order to accomplish this task we need to enable

zone bandwidth control on the gatekeeper:

HQ_Rtr(config)#gatekeeper

HQ_Rtr(config-gk)#bandwidth total zone CME 48

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Page 852: Ccie Voice Advanced Lab Workbook

High Availability Features

1. In order to accomplish this task we need to enable

“call-manager-fallback” in the BR1 gateway:

application

global

service alternate DEFAULT

call-manager-fallback

secondary-dialtone 9

max-conferences 4 gain -6

ip source-address 10.1.130.1 port 2000

max-ephones 2

max-dn 10

transfer-pattern .T

voicemail 917029461170

call-forward pattern .T

2. This task was accomplished earlier by setting the BR1

phone 1 device pool to “BR1-NOSRST” which had the SRST

reference disabled.

3. This was completed during the call routing section as

the dial-peers were created with 9 at the beginning.

4. In order to accomplish this we need to add the proper

COR list to the DN of BR1 phone 2 in the incoming

direction:

call-manager-fallback

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Page 853: Ccie Voice Advanced Lab Workbook

cor incoming Phone2 1 2002

5. Most of this task has already been completed during

the phone setup section. We created the AAR groups

with the appropriate prefixes and then assigned the

DN’s to the appropriate AAR group. The last step is to

enable the AAR service via the callmanager service

parameters:

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Page 854: Ccie Voice Advanced Lab Workbook

Media Resources Management

1. In order to accomplish this task we first need to

create the resource in callmanager:

Next we need to configure the SCCP and DSPFARM

settings on the HQ router:

sccp local Loopback0

sccp ccm 10.1.200.21 identifier 2 priority 2 version 7.0

sccp ccm 10.1.200.25 identifier 1 priority 1 version 7.0

sccp

!

sccp ccm group 1

associate ccm 1 priority 1

associate ccm 2 priority 2

associate profile 2 register HQTRANS

associate profile 1 register HQCONF

!

dspfarm profile 2 transcode

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Page 855: Ccie Voice Advanced Lab Workbook

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

maximum sessions 4

associate application SCCP

!

dspfarm profile 1 conference

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

codec g729r8

codec g729br8

maximum sessions 4

associate application SCCP

!Tip - You can use the “show sccp” and the show “dspfarm profile” command

to verify gateway configuration. You may need to disable and then enable the

SCCP function if the resources aren’t registering.

2. In order to accomplish this task we first need to

create the resource in callmanager:

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Time Saver - This is another great instance of where copy and paste can

help you as the configuration will be very similar to the HQ router.

sccp local Loopback0

sccp ccm 10.1.200.25 identifier 1 priority 1 version 7.0

sccp ccm 10.1.200.21 identifier 2 priority 2 version 7.0

sccp

!

sccp ccm group 1

associate ccm 1 priority 1

associate ccm 2 priority 2

associate profile 1 register BR1CONF

associate profile 2 register BR1TRANS

!

dspfarm profile 2 transcode

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

codec gsmfr

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maximum sessions 4

associate application SCCP

!

dspfarm profile 1 conference

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

codec g729r8

codec g729br8

maximum sessions 4

associate application SCCP

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3. To accomplish this task we first need to create the

resource in callmanager:

The gateway configuration portion was accomplished

above in the conference bridge section.

4. To accomplish this task we first need to create the

resource in callmanager:

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The gateway configuration portion was accomplished

above in the conference bridge section.

5. In a previous step we create all the media resource

groups and list. Now we can add the proper media

resources to the proper groups. The publisher MOH

server will need to be put into the “PUB_MOH” media

resource group and we will need to make sure that this

is the MOH only server listed in the media resource

list for HQ.

6. We need to perform the same actions for BR1 but with

the subscriber server being put into the “SUB_MOH”

media resource group and making sure that this is the

only MOH server listed in the media resource list for

BR1. In addition we will need to enable the default

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audio source, subscriber MOH server, and the “SUB_MOH”

media resource group for multicasting:

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We need to insure that the “Max Hops” is set to “1” so

that the multicast stream stays local.

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Now we need to enable the G.729 codec through the

Cisco IP Voice Media Streaming App service parameters:

We then need to configure the BR1 gateway to support

multicast and the router to play the MOH file from

flash:

call-manager-fallback

moh music-on-hold.au

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multicast moh 239.1.1.1 port 16388 route 10.1.130.1

ip multicast-routing

interface Vlan130

ip pim sparse-dense-mode

! Tip - To test place a call from HQ to BR1 and then put the call on hold from

both ends. You should receive MOH but hear different music on both sides.

7. In order to accomplish this we first need to create

the MeetMe conference resources in callmanager:

We then need to change the callmanager service

parameter to allow for 10 participants per conference:

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8. This is accomplished through the same callmanager

service parameter section as the task above by setting

the “Drop Ad Hoc Conference” parameter correctly.

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QoS Features

1. We can accomplish this with the following

configuration.

On the HQ gateway:

interface Serial0/2/0.1 point-to-point

bandwidth 384

frame-relay interface-dlci 201

auto qos voip trust fr-atm

On the BR1 Gateway:

interface Serial0/2/0.1 point-to-point

bandwidth 384

frame-relay interface-dlci 101 ppp

auto qos voip trust fr-atm

The on both gateways modify the auto generated policy

map per the requirements:

policy-map AutoQoS-Policy-Trust

class AutoQoS-VoIP-RTP-Trust

priority percent 33

class AutoQoS-VoIP-Control-Trust

bandwidth percent 5

class class-default

fair-queue

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2. We can accomplish this with the following

configuration.

On the HQ gateway:

interface Serial0/2/0.2 point-to-point

bandwidth 1200

frame-relay interface-dlci 202

auto qos voip trust

On the BR2 gateway:

interface Serial0/2/0.1 point-to-point

bandwidth 1200

frame-relay interface-dlci 102

auto qos voip trust

Then on both gateways modify the auto generated policy

map per the requirements:

policy-map AutoQoS-Policy-Trust2

class AutoQoS-VoIP-RTP-Trust

priority 360

class AutoQoS-VoIP-Control-Trust

bandwidth 60

class class-default

fair-queue

3. This was accomplished earlier under the two dial-peers

pointing traffic back to the callmanagers.

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Application and End Users

In order to move forward past this section we need to

define all the users. Outlined below in tables are the

users that need to be created and the settings for those

users.

!Tip - Pay close attention to the associated devices and rights of each user.

Before you start to add user you will need to create a “Standard AXL User”

group and assign the “Standard AXL API Access” role to that group.

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!""#$%&'$()*+,-.*

+,-.*/0* /)1(.2&'$()*

!"#$%&&

'(()#*+"#,-&./01&2-3,14+"#,-&

510/0-*0&$1,6(& 7"+-8+18&510/0-*0&91,6(&

'**0("&510/0-*0&76:/*1#("#,-& ;&

'**0("&<6"=,3=8#+),9&>?@?>& ;&

'**0("&.-/,)#*#"08&;,"#3#*+"#,-& ;&

'**0("&>0()+*0/&A0+801& ;&

B0C#*0&2-3,14+"#,-&

!,-"1,))08&B0C#*0/& 7?5DDEF!GHBIB!!&7?5DDIDJK!HGLFK&7?5DDIDJK!HB'IL&7?5DDIDJK!IB!@J&

!'5@&2-3,14+"#,-&

'//,*#+"08&!'5@&51,3#)0/& &

5014#//#,-/&2-3,14+"#,-&

$1,6(/& 7"+-8+18&!M2&')),%&!,-"1,)&,3&'))&B0C#*0/&7"+-8+18&!M2&?-+:)08&

>,)0/& 7"+-8+18&!M2&')),%&!,-"1,)&,3&'))&B0C#*0/&7"+-8+18&!M2&?-+:)08&

&&

25!!&&

'(()#*+"#,-&./01&2-3,14+"#,-&

510/0-*0&$1,6(& 7"+-8+18&510/0-*0&91,6(&

'**0("&510/0-*0&76:/*1#("#,-& ;&

'**0("&<6"=,3=8#+),9&>?@?>& ;&

'**0("&.-/,)#*#"08&;,"#3#*+"#,-& ;&

'**0("&>0()+*0/&A0+801& ;&

B0C#*0&2-3,14+"#,-&

!,-"1,))08&B0C#*0/& &

!'5@&2-3,14+"#,-&

'//,*#+"08&!'5@&51,3#)0/& &

5014#//#,-/&2-3,14+"#,-&

$1,6(/& 7"+-8+18&'NO&./01&

>,)0/& 7"+-8+18&'NO&'52&'**0//&

&&

.-#"P&&

'(()#*+"#,-&./01&2-3,14+"#,-&

510/0-*0&$1,6(& 7"+-8+18&510/0-*0&91,6(&

'**0("&510/0-*0&76:/*1#("#,-& ;&

'**0("&<6"=,3=8#+),9&>?@?>& ;&

'**0("&.-/,)#*#"08&;,"#3#*+"#,-& ;&

'**0("&>0()+*0/&A0+801& ;&

B0C#*0&2-3,14+"#,-&

!,-"1,))08&B0C#*0/& &

!'5@&2-3,14+"#,-&

'//,*#+"08&!'5@&51,3#)0/& &

5014#//#,-/&2-3,14+"#,-&

$1,6(/& 7"+-8+18&'NO&./01&

>,)0/& 7"+-8+18&'NO&'52&'**0//&

&

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The next list of Application users is created during the

setup of the contact center server and therefore will not

need to be create:

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!""#$%&'$()*+,-.*

+,-.*/0* /)1(.2&'$()*

!

"#$%&'('!!

)%%*&+$#&,-!./01!2-3,14$#&,-!

510/0-+0!61,7%! 8#$-9$19!510/0-+0!:1,7%!

)++0%#!510/0-+0!87;/+1&%#&,-! <!

)++0%#!=7#>,3>9&$*,:!?@A@?! <!

)++0%#!.-/,*&+&#09!<,#&3&+$#&,-! <!

)++0%#!?0%*$+0/!B0$901! <!

C0D&+0!2-3,14$#&,-!

E,-#1,**09!C0D&+0/! ))('F'G!))('F'H!))('F'I!))('F'J!))('F'K!))!2EC!

E)5A!2-3,14$#&,-!

)//,+&$#09!E)5A!51,3&*0/! !

5014&//&,-/!2-3,14$#&,-!

61,7%/! 8#$-9$19!EL2!@-$;*09!

?,*0/! 8#$-9$19!EL2!@-$;*09!

!!

14"#$%&'!!

)%%*&+$#&,-!./01!2-3,14$#&,-!

510/0-+0!61,7%! 8#$-9$19!510/0-+0!:1,7%!

)++0%#!510/0-+0!87;/+1&%#&,-! <!

)++0%#!=7#>,3>9&$*,:!?@A@?! <!

)++0%#!.-/,*&+&#09!<,#&3&+$#&,-! <!

)++0%#!?0%*$+0/!B0$901! <!

C0D&+0!2-3,14$#&,-!

E,-#1,**09!C0D&+0/! 8@5FF'HEIMCNCEE!8@5FFNFKOEMIJHO!8@5FFNFKOEMC)NJ!8@5FFNFKOENCEAK!

E)5A!2-3,14$#&,-!

)//,+&$#09!E)5A!51,3&*0/! !

5014&//&,-/!2-3,14$#&,-!

61,7%/! 8#$-9$19!EL2!@-$;*09!

?,*0/! 8#$-9$19!EL2!@-$;*09!

!!

!

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The following End Users will also need to be created with

the correct settings from the table:

!"#$%&'($)*+"#'"&'#,$

%&'($-.$ /'0'12+"'$

3456'($

*+"7(+00'#$

.'89:'&$

*+"7(+00'#$

;(+<90'&$

=&&+:9>7'#$

!?7'"&9+"&$

@(+41&$ A+0'&$

!"#$ %&&#$ '()&&*&+,-*.-/+$ $$ %&&#$01$)23!"#341567189$%&&#$3$4)--$

'581:87:$--;$(1:$<=67=$'581:87:$-24$(18>96:$

'581:87:$--;$(1:$<=67=$'581:87:$--;<'("$?:@010=57850A1$'581:87:$-24$(18>96:$

!"%$ %&&%$ '()&&#B-C%.*.--$ $$ %&&%$01$)23!"#341567189$

'581:87:$--;$(1:$<=67=$'581:87:$-24$(18>96:$

'581:87:$--;$(1:$<=67=$'581:87:$--;<'("$?:@010=57850A1$'581:87:$-24$(18>96:$

DE#$ #&&#$ '()&&*&+,-%CFB,$<)-DE#$

$$ #&&#$01$)23DE341567189$#&&#$3$4)--$

'581:87:$--;$(1:$<=67=$'581:87:$-24$(18>96:$

'581:87:$--;$(1:$<=67=$'581:87:$--;<'("$?:@010=57850A1$'581:87:$-24$(18>96:$

DE%$ #&&%$ '()&&*&+,-%.?*F$ $$ #&&%$01$)23DE341567189$

'581:87:$--;$(1:$<=67=$'581:87:$-24$(18>96:$

'581:87:$--;$(1:$<=67=$'581:87:$--;<'("$?:@010=57850A1$'581:87:$-24$(18>96:$

G7=8:@01$ $$ $$ $$ H$IA16$J$ $$ $$

KAL1$ #&&C$ $$ MAL1$ #&&C$01$)23DE341567189$

'581:87:$--;$(1:$<=67=$'581:87:$-24$(18>96:$

'581:87:$--;$(1:$<=67=$'581:87:$--;<'("$?:@010=57850A1$'581:87:$-24$(18>96:$

!

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Voicemail Integrations

1. In order to accomplish this we first need to perform

all the required setup in callmanager for voicemail.

We start first with creating the voicemail ports:

Time Saver - You can create the ports manually or through the wizard.

Obviously the wizard is designed to save time as it creates the line group

for you will all the voicemail ports as members.

Below is the line group configuration:

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Next you will need to add this line group to a hunt list:

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Now we need to create the voicemail pilot (Hunt Pilot)

and point it to the correct hunt list:

We can now create the MWI on and MWI off extensions:

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We can now set the default voicemail pilot and

profile:

!Tip - You will see that we create a profile named strip. This will be used later

in task 5 in order to strip the “*” out of the extension before it rolls to

voicemail. The settings of this profile are shown below.

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Now in order to start the integration with Unity

Connections we will first need to create a port group:

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Next will need to define the callmanager servers Unity

will use through the Edit->Servers page under the port

group:

During the port group creation we were asked how many

ports and the device name prefix. This created the

ports for us automatically:

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Now we can finish the integration by settings the

parameters for the phone system to include the AXL

servers for importing users from callmanager:

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! Tip - This is also where the loop detection is enabled which is required in the

next task.

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Now we can define the AXL servers through the Edit->

Cisco Unified Communications Manager AXl Servers under

the phone system:

!Note - You will see the application user “Unity” which we create previously.

This user must be in the “Standard AXL User Group” that we created in order

to import users from callmanager.

2. In Unity Connections loop prevention is on by default.

However, you should be aware that there are two ways to

accomplish loop prevention. One is by extension and the

other is through DTMF tone based recognition.

3. In order to accomplish this task you must first modify

the default voicemail user template to change the default

password. You can then import all the users into Unity:

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You can now go to the import users page an import all

users:

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When finished your users page should look like the

following:

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! Tip - If the primary extension of the user is not set then the user will not show

up for import.

4. In order to configure the auto attendant we will first

need to create the appropriate CTI route point in

callmanager and forward all calls to voicemail for this

CTI route point:

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Time Saver -You can see two CTI route points listed as the second one will

be used to record the greeting for the call handler in this task.

We can setup the Greetings Administrator access for HQ

phone 1 user through the call routing section:

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Once this is complete we can now setup the call

handler for the auto attendant:

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Next we can setup the caller input options as required

in the task:

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We now need to enable and setup the closed extension

to take a message:

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We also need to modify the standard greeting so that a

message is not taken:

We need to also assign the call handler owner so that

the HQ phone 1 user can access the call handler and

change the greeting via the Greetings Administrator:

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5. Part of this task has already been configured during the

phone setup section and callmanager voicemail section.

What we need to double check is that the voicemail

profile named “STRIP” is assigned to the BR1 phone 2

extension of “*2002”:

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Page 896: Ccie Voice Advanced Lab Workbook

6. In order to accomplish this task we first need to define

access to the CUE module, the ephone-dn’s for MWI, and

then dial-peers for the correct number(s) that need to be

forwarded to CUE:

interface Service-Engine1/0

ip unnumbered FastEthernet0/0.230

service-module ip address 10.1.230.2 255.255.255.0

service-module ip default-gateway 10.1.230.1

ip route 10.1.230.2 255.255.255.255 Service-Engine1/0

ephone-dn 4

number 8001....

mwi on

ephone-dn 5

number 8002....

mwi off

dial-peer voice 3180 voip

destination-pattern 3180

session protocol sipv2

session target ipv4:10.1.230.2

dtmf-relay rtp-nte

codec g711ulaw

no vad

!

dial-peer voice 3000 voip

destination-pattern 3000

session protocol sipv2

session target ipv4:10.1.230.2

dtmf-relay rtp-nte

codec g711ulaw

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no vad

Time Saver - In order to save time we can go ahed and create the dial-peer

need for task 8.

You can now start the CUE initialization process by

session into the CUE module from the BR2 gateway with

the following command:

BR2#service-module service-engine 1/0 session

Once you have completed the setup the CUE module will

reboot and is ready for configuration via the web

interface or via command line. You will now need to

set the MWI on and MWI off:

ccn application ciscomwiapplication aa description "ciscomwiapplication" enabled maxsessions 8 script "setmwi.aef" parameter "CallControlGroupID" "0" parameter "strMWI_OFF_DN" "8002" parameter "strMWI_ON_DN" "8001" end application

You can test access by pressing the voicemail button

on any BR2 phone.

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Page 898: Ccie Voice Advanced Lab Workbook

7. You can accomplish this via the web interface or via the

command line. We will show through the command line:

username br2phone3 create username BR2Phone2 create username BR2Phone1 create username admin create

username BR2Phone2 phonenumber "3002"

username BR2Phone1 phonenumber "3001"

username BR2Phone2 phonenumberE164 "442076303002"

username BR2Phone1 phonenumberE164 "442076303001"

voicemail mailbox owner "BR2Phone1" size 21176 description "BR2Phone1 mailbox" end mailbox voicemail mailbox owner "BR2Phone2" size 21176 description "BR2Phone2 mailbox" end mailbox

! Tip - You can test access and MWI by leaving a message and making sure the

MWI is working for each mailbox.

8. Part of this task was completed in task 6 and now we just

need to define the auto attendant in CUE:

ccn trigger sip phonenumber 3000

application "autoattendant"

enabled

maxsessions 8

end trigger

9. The was already accomplished earlier during the call

routing section. You will just need to test!

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Page 900: Ccie Voice Advanced Lab Workbook

IPCC Express Voice Application

1. In order to accomplish this task you will need to

complete the normal IPCC Express setup. Once this is

done you will need to create a Call Control Group

based on the task requirements:

!Tip - If required you can click the “Show More...” button and set the partition,

calling search space, AAR group, and other settings that may be required. In

this case we would probably want to set things like the device pool and

location.

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2. To accomplish this task you will need to create a new

application and trigger using the default auto

attendant script:

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3. First we need to create the application and trigger:

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We now can modify the resources and assign the skills

required for task 5:

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! Tip - If you don’t see any resources in IPCC Express then you probably

haven’t assigned the users an IPCC extension in callmanager.

We now need to create the contact services queue with

the same name we referenced under the application:

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We now need to create the IP Agent Phone Service:

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Page 908: Ccie Voice Advanced Lab Workbook

We can then assign the IP phone service to the proper

phones:

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Page 909: Ccie Voice Advanced Lab Workbook

You should now be able to login to the IP Agent Phone

service and test calls into the queue.

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Page 910: Ccie Voice Advanced Lab Workbook

4. In order to accomplish this we can modify the default

script to place the call on hold and then off hold

during the queuing process:

5. This was accomplished in task 3 by assigning the users

skills and then modifying the queue to route based on

“Most Skilled”.

6. This is shown in the script example provided under

task 4.

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Page 912: Ccie Voice Advanced Lab Workbook

7. In order to accomplish this we need to write out the

calling number to a variable then perform a check on

that number. If matched we need to set the priority of

the call:

! Tip - You should get to know the “Get Reporting Statistics” function in IPCC

Express. The function will allow you to create checks for just about any task.

! Tip - Make sure anytime that you modify a script that you refresh the script

and associated application for the changes to take affect.

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CallManager Voice Application

1. Part of this task was already completed when we created

the user in callmanager. We now need to create the IP

phone service:

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Next we can create the device profile:

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We need to make sure we subscribe the device profile

to the extension mobility phone service:

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We then need to enable the required phones for extension

mobility and subscribe them to the IP phone service for

extension mobility:

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Presence

1. The first step is to run through the setup for

Presence after the initial install. Once this is

complete we can setup the required SIP trunk in

callmanager and set the SIP publish trunk in the

callmanager service parameters:

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Page 918: Ccie Voice Advanced Lab Workbook

Now set the CUP Publish Trunk in the callmanager

service parameters:

We can now create the CUPC in callmanager:

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Make sure on all the phones and CUPC in callmanager

that the subscribe calling search space is set

accordingly:

Next we need to assign the capabilities in

callmanager:

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Page 920: Ccie Voice Advanced Lab Workbook

Once this is complete we now need to finish the

configuration of the presence server. We can start by

validating the setup via System->CUCM Publisher:

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Page 921: Ccie Voice Advanced Lab Workbook

You can now configure the incoming and outgoing ACL’s:

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We now need to configure the CM SIP Publish trunk

under the Presence->Settings:

We now need to configure the callmanager as a presence

gateway:

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Page 923: Ccie Voice Advanced Lab Workbook

We now need to modify the proxy services settings:

We can then start configuring the CUPC settings:

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Page 924: Ccie Voice Advanced Lab Workbook

Next we need to configure the Unity server as a

voicemail server and then configure the voicemail

profile:

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We can now set the proper profiles for the HQ phone 1

user:

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Page 927: Ccie Voice Advanced Lab Workbook

Next we need to configure the “Deskphone Control

Settings” so that the CUPC client can control the

phone:

! Tip - The phones that need to be controlled by the “CtiGw” user were defined

earlier during the Application User section.

! Tip - You will also want to make sure that the phone being controlled has an

owner assigned and is associated to the end user.

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We can then enable the HQ phone 1 user for desktop

phone control:

Next you will need to the proxy domain and restart the

presence and proxy services:

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!Tip - You will need to create a host file entry on the machine running CUPC

based on the domain name you select in the presence server parameters.

Example: “presence.ccbootcamp.com” pointing to 10.1.200.23.

2. We need to first enable Unified Client access within

Unity Connections:

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Page 930: Ccie Voice Advanced Lab Workbook

Next we need to setup the Unity account user name and

password in the CUPC client and check that all

services are up and running:

!Note - You can switch between modes and check services for each setup. Keep

in mind that the service for the mode your not in will show as down or not

configured.

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