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CCBOOTCAMPs

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
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CCBOOTCAMPs 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
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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
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Systems, Inc. The contents contained wherein, are not associated with
or endorsed by Cisco Systems, Inc.
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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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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.com
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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, CCBOOTCAMPs 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 youre
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 #18065
CCIE Voice Program Manager
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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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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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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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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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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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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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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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4. Configure BR1 phones to have IP addresses of 130140
using the IOS DHCP on the BR1 Router.
5. Configure BR2 phones to have IP addresses 210220
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.
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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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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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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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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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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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:
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 pins
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.
For questions: www.voiceie.com
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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.
For questions: www.voiceie.com
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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.
For questions: www.voiceie.com
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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 - Dont 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 - Dont 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 - Dont 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 - Dont forget to set the time zone settings in CallManager as it may not be
mentioned in the phone setup tasks.
For questions: www.voiceie.com
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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 dont 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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Calling Search Space
Name Description Route Partitions for this Calling Search Space
CSS_BR1_AAR CSS_BR1_AAR PT-BR1-AAR
CSS_BR1_INT CSS_BR1_INT PT-BR1-911
PT-BR1-Internal
PT-BR1-TOLLBYPASS
PT-HQ-Internal
PT-BR2-Internal
PT-BR1-Local
PT-BR1-LD
PT-BR1-INT
CSS_BR1_INTERNAL CSS_BR1_INTERNAL PT-BR1-911
PT-BR1-Internal
PT-BR1-TOLLBYPASS
PT-HQ-Internal
PT-BR2-Internal
CSS_BR1_LD CSS_BR1_LD PT-BR1-911
PT-BR1-Internal
PT-BR1-TOLLBYPASS
PT-HQ-Internal
PT-BR2-Internal
PT-BR1-Local
PT-BR1-LD
CSS_BR1_LOCAL CSS_BR1_LOCAL PT-BR1-911
PT-BR1-Internal
PT-BR1-TOLLBYPASS
PT-HQ-Internal
PT-BR2-Internal
PT-BR1-Local
CSS_BR2 CSS_BR2 PT-BR1-Internal
PT-BR2-Internal
PT-HQ-Internal
PT-BR2-TOLLBYPASS
CSS_HQ_AAR CSS_HQ_AAR PT-HQ-AAR
CSS_HQ_INT CSS_HQ_INT PT-HQ-911
PT-HQ-Internal
PT-HQ-TOLLBYPASS
PT-BR1-Internal
PT-BR2-Internal
PT-HQ-Local
PT-HQ-LD
PT-HQ-INT
CSS_HQ_INTERNAL CSS_HQ_INTERNAL PT-HQ-911
PT-HQ-Internal
PT-HQ-TOLLBYPASS
PT-BR1-Internal
PT-BR2-Internal
CSS_HQ_LD CSS_HQ_LD PT-HQ-911
PT-HQ-Internal
PT-HQ-TOLLBYPASS
PT-BR1-Internal
PT-BR2-Internal
PT-HQ-Local
PT-HQ-LD
CSS_HQ_LOCAL CSS_HQ_LOCAL PT-HQ-911
PT-HQ-Internal
PT-HQ-TOLLBYPASS
PT-BR1-Internal
PT-BR2-Internal
PT-HQ-Local
CSS_VOICEMAIL CSS_VOICEMAIL PT-BR1-Internal
PT-HQ-Internal


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 DNs:
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 DNs:
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 DNs.
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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Route List
Route List associated Route Group with members
BR1 AAR
Description BR1 AAR
CUCM Group SUB-PUB
Enabled Y


BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits < None >
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members 10.10.32.2 (All Ports)



BR1_911
Description BR1_911
CUCM Group SUB-PUB
Enabled Y


BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members 10.10.32.2 (All Ports)



BR1_BR2
Description BR1_BR2
CUCM Group SUB-PUB
Enabled Y


BR2_TRUNK
Calling Party Transformations
Use Calling Party's External Mask Off
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
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Route List
Route List associated Route Group with members
Prefix Digits (Outgoing Calls)
Members BR2 (All Ports)

BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 901144207630
Members 10.10.32.2 (All Ports)



BR1_BR2_TOLLBYPASS
Description BR1_BR2_TOLLBYPASS
CUCM Group SUB-PUB
Enabled Y


BR2_TRUNK
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 9
Members BR2 (All Ports)

BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
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Route List
Route List associated Route Group with members
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 901144207
Members 10.10.32.2 (All Ports)



BR1_HQ_TOLLBYPASS
Description BR1_HQ_TOLLBYPASS
CUCM Group SUB-PUB
Enabled Y


HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)

BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 91702
Members 10.10.32.2 (All Ports)



BR1_INTERNATIONAL BR1_GW
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Route List
Route List associated Route Group with members
Description BR1_INTERNATIONAL
CUCM Group SUB-PUB
Enabled Y


Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members 10.10.32.2 (All Ports)



BR1_LD
Description BR1_LD
CUCM Group SUB-PUB
Enabled Y


BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits < None >
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members 10.10.32.2 (All Ports)

HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)
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Route List
Route List associated Route Group with members



BR1_LOCAL
Description BR1_LOCAL
CUCM Group SUB-PUB
Enabled Y


BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits < None >
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members 10.10.32.2 (All Ports)

HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 1702
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)



HQ AAR
Description HQ AAR
CUCM Group SUB-PUB
Enabled Y


HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
For questions: www.voiceie.com
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Route List
Route List associated Route Group with members
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)



HQ_911
Description HQ_911
CUCM Group SUB-PUB
Enabled Y


HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)



HQ_BR1_TOLLBYPASS
Description HQ_BR1_TOLLBYPASS
CUCM Group SUB-PUB
Enabled Y


BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 9
Members 10.10.32.2 (All Ports)

HQ_GW
Calling Party Transformations
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Route List
Route List associated Route Group with members
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 1702
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)



HQ_BR2
Description HQ_BR2
CUCM Group SUB-PUB
Enabled Y


BR2_TRUNK
Calling Party Transformations
Use Calling Party's External Mask Off
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members BR2 (All Ports)

HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 01144207630
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)


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Route List
Route List associated Route Group with members

HQ_BR2_TOLLBYPASS
Description HQ_BR2_TOLLBYPASS
CUCM Group SUB-PUB
Enabled Y


BR2_TRUNK
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 9
Members BR2 (All Ports)

HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 01144207
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)



HQ_INTERNATIONAL
Description HQ_INTERNATIONAL
CUCM Group SUB-PUB
Enabled Y


HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
For questions: www.voiceie.com
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Route List
Route List associated Route Group with members
Prefix Digits (Outgoing Calls)
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)



HQ_LD
Description HQ_LD
CUCM Group SUB-PUB
Enabled Y


HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)

BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 9
Members 10.10.32.2 (All Ports)



HQ_LOCAL
Description HQ_LOCAL
CUCM Group SUB-PUB
Enabled Y

HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
For questions: www.voiceie.com
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Route List
Route List associated Route Group with members
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)

BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 91702
Members 10.10.32.2 (All Ports)


For questions: www.voiceie.com


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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
DNs 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 arent 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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Application User
User ID Information
CtiGw

Application User Information
Presence Group Standard Presence group
Accept Presence Subscription N
Accept Out-of-dialog REFER N
Accept Unsolicited Notification N
Accept Replaces Header N
Device Information
Controlled Devices SEP0016C72D3DCC
SEP003094C27864
SEP003094C2DA38
SEP003094C3DCF9
CAPF Information
Associated CAPF Profiles
Permissions Information
Groups Standard CTI Allow Control of All Devices
Standard CTI Enabled
Roles Standard CTI Allow Control of All Devices
Standard CTI Enabled


IPCC

Application User Information
Presence Group Standard Presence group
Accept Presence Subscription N
Accept Out-of-dialog REFER N
Accept Unsolicited Notification N
Accept Replaces Header N
Device Information
Controlled Devices
CAPF Information
Associated CAPF Profiles
Permissions Information
Groups Standard AXL User
Roles Standard AXL API Access


Unity

Application User Information
Presence Group Standard Presence group
Accept Presence Subscription N
Accept Out-of-dialog REFER N
Accept Unsolicited Notification N
Accept Replaces Header N
Device Information
Controlled Devices
CAPF Information
Associated CAPF Profiles
Permissions Information
Groups Standard AXL User
Roles Standard AXL API Access

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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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Application User
User ID Information

jtapi1_1

Application User Information
Presence Group Standard Presence group
Accept Presence Subscription N
Accept Out-of-dialog REFER N
Accept Unsolicited Notification N
Accept Replaces Header N
Device Information
Controlled Devices AA_1015
AA_1016
AA_1017
AA_1018
AA_1019
AA
ICD
CAPF Information
Associated CAPF Profiles
Permissions Information
Groups Standard CTI Enabled
Roles Standard CTI Enabled


rmjtapi1

Application User Information
Presence Group Standard Presence group
Accept Presence Subscription N
Accept Out-of-dialog REFER N
Accept Unsolicited Notification N
Accept Replaces Header N
Device Information
Controlled Devices SEP0016C72D3DCC
SEP003094C27864
SEP003094C2DA38
SEP003094C3DCF9
CAPF Information
Associated CAPF Profiles
Permissions Information
Groups Standard CTI Enabled
Roles Standard CTI Enabled

For questions: www.voiceie.com


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The following End Users will also need to be created with
the correct settings from the table:



End User (Condensed)
User ID Telephone
Number
Controlled
Devices
Controlled
Profiles
Associated
Extensions
Groups Roles
BR1 2001 SEP003094C3DCF9 2001 in PT-
BR1-Internal
2001 - IPCC
Standard CCM End
Users
Standard CTI
Enabled
Standard CCM End Users
Standard CCMUSER
Administration
Standard CTI Enabled
BR2 2002 SEP0016C72D3DCC 2002 in PT-
BR1-Internal
Standard CCM End
Users
Standard CTI
Enabled
Standard CCM End Users
Standard CCMUSER
Administration
Standard CTI Enabled
HQ1 1001 SEP003094C27864
UPCHQ1
1001 in PT-
HQ-Internal
1001 - IPCC
Standard CCM End
Users
Standard CTI
Enabled
Standard CCM End Users
Standard CCMUSER
Administration
Standard CTI Enabled
HQ2 1002 SEP003094C2DA38 1002 in PT-
HQ-Internal
Standard CCM End
Users
Standard CTI
Enabled
Standard CCM End Users
Standard CCMUSER
Administration
Standard CTI Enabled
crsadmin < None >
john 1007 John 1007 in PT-
HQ-Internal
Standard CCM End
Users
Standard CTI
Enabled
Standard CCM End Users
Standard CCMUSER
Administration
Standard CTI Enabled

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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 conguration:
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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 dont see any resources in IPCC Express then you probably
havent 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 ACLs:
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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 130140
using the IOS DHCP on the BR1 Router.
5. Configure BR2 phones to have IP addresses 210220
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 pins
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 - Dont 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 - Dont 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 - Dont 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 - Dont 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 dont 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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Calling Search Space
Name Description Route Partitions for this Calling Search Space
CSS_BR1_AAR CSS_BR1_AAR PT-BR1-AAR
CSS_BR1_INT CSS_BR1_INT PT-BR1-911
PT-BR1-Internal
PT-BR1-TOLLBYPASS
PT-HQ-Internal
PT-BR2-Internal
PT-BR1-Local
PT-BR1-LD
PT-BR1-INT
CSS_BR1_INTERNAL CSS_BR1_INTERNAL PT-BR1-911
PT-BR1-Internal
PT-BR1-TOLLBYPASS
PT-HQ-Internal
PT-BR2-Internal
CSS_BR1_LD CSS_BR1_LD PT-BR1-911
PT-BR1-Internal
PT-BR1-TOLLBYPASS
PT-HQ-Internal
PT-BR2-Internal
PT-BR1-Local
PT-BR1-LD
CSS_BR1_LOCAL CSS_BR1_LOCAL PT-BR1-911
PT-BR1-Internal
PT-BR1-TOLLBYPASS
PT-HQ-Internal
PT-BR2-Internal
PT-BR1-Local
CSS_BR2 CSS_BR2 PT-BR1-Internal
PT-BR2-Internal
PT-HQ-Internal
PT-BR2-TOLLBYPASS
CSS_HQ_AAR CSS_HQ_AAR PT-HQ-AAR
CSS_HQ_INT CSS_HQ_INT PT-HQ-911
PT-HQ-Internal
PT-HQ-TOLLBYPASS
PT-BR1-Internal
PT-BR2-Internal
PT-HQ-Local
PT-HQ-LD
PT-HQ-INT
CSS_HQ_INTERNAL CSS_HQ_INTERNAL PT-HQ-911
PT-HQ-Internal
PT-HQ-TOLLBYPASS
PT-BR1-Internal
PT-BR2-Internal
CSS_HQ_LD CSS_HQ_LD PT-HQ-911
PT-HQ-Internal
PT-HQ-TOLLBYPASS
PT-BR1-Internal
PT-BR2-Internal
PT-HQ-Local
PT-HQ-LD
CSS_HQ_LOCAL CSS_HQ_LOCAL PT-HQ-911
PT-HQ-Internal
PT-HQ-TOLLBYPASS
PT-BR1-Internal
PT-BR2-Internal
PT-HQ-Local
CSS_VOICEMAIL CSS_VOICEMAIL PT-BR1-Internal
PT-HQ-Internal


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 DNs:
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 DNs:
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 DNs.
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 DNs 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.
For questions: www.voiceie.com
v.a.l.03.09.05.cf.r05.24.06.pages
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Copyright 2009, Network Learning, Incorporated
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.
For questions: www.voiceie.com
v.a.l.03.09.05.cf.r05.24.06.pages
257
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Next we will need to configure the gateway in
CallManager:
For questions: www.voiceie.com
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For questions: www.voiceie.com
v.a.l.03.09.05.cf.r05.24.06.pages
259
www.ccbootcamp.com Toll Free 877.654.2243 sales@ccbootcamp.com
Copyright 2009, Network Learning, Incorporated
For questions: www.voiceie.com
v.a.l.03.09.05.cf.r05.24.06.pages
260
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Copyright 2009, Network Learning, Incorporated
!
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.
For questions: www.voiceie.com
v.a.l.03.09.05.cf.r05.24.06.pages
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Copyright 2009, Network Learning, Incorporated
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
For questions: www.voiceie.com
v.a.l.03.09.05.cf.r05.24.06.pages
262
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Copyright 2009, Network Learning, Incorporated
Next we will need to setup BR1 as an H.323 gateway in
CallManager:
For questions: www.voiceie.com
v.a.l.03.09.05.cf.r05.24.06.pages
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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!
For questions: www.voiceie.com
v.a.l.03.09.05.cf.r05.24.06.pages
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Copyright 2009, Network Learning, Incorporated
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
For questions: www.voiceie.com
v.a.l.03.09.05.cf.r05.24.06.pages
265
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Copyright 2009, Network Learning, Incorporated
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.
For questions: www.voiceie.com
v.a.l.03.09.05.cf.r05.24.06.pages
266
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Now we need to add the trunk:
For questions: www.voiceie.com
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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#
For questions: www.voiceie.com
v.a.l.03.09.05.cf.r05.24.06.pages
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Copyright 2009, Network Learning, Incorporated
For questions: www.voiceie.com
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Copyright 2009, Network Learning, Incorporated
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.
For questions: www.voiceie.com
v.a.l.03.09.05.cf.r05.24.06.pages
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For questions: www.voiceie.com
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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
For questions: www.voiceie.com
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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:
For questions: www.voiceie.com
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For questions: www.voiceie.com
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Then next the route lists in the following tables with
the proper settings:
For questions: www.voiceie.com
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Route List
Route List associated Route Group with members
BR1 AAR
Description BR1 AAR
CUCM Group SUB-PUB
Enabled Y


BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits < None >
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members 10.10.32.2 (All Ports)



BR1_911
Description BR1_911
CUCM Group SUB-PUB
Enabled Y


BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members 10.10.32.2 (All Ports)



BR1_BR2
Description BR1_BR2
CUCM Group SUB-PUB
Enabled Y


BR2_TRUNK
Calling Party Transformations
Use Calling Party's External Mask Off
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
For questions: www.voiceie.com
v.a.l.03.09.05.cf.r05.24.06.pages
276
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Copyright 2009, Network Learning, Incorporated


Route List
Route List associated Route Group with members
Prefix Digits (Outgoing Calls)
Members BR2 (All Ports)

BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 901144207630
Members 10.10.32.2 (All Ports)



BR1_BR2_TOLLBYPASS
Description BR1_BR2_TOLLBYPASS
CUCM Group SUB-PUB
Enabled Y


BR2_TRUNK
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 9
Members BR2 (All Ports)

BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
For questions: www.voiceie.com
v.a.l.03.09.05.cf.r05.24.06.pages
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Copyright 2009, Network Learning, Incorporated


Route List
Route List associated Route Group with members
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 901144207
Members 10.10.32.2 (All Ports)



BR1_HQ_TOLLBYPASS
Description BR1_HQ_TOLLBYPASS
CUCM Group SUB-PUB
Enabled Y


HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)

BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 91702
Members 10.10.32.2 (All Ports)



BR1_INTERNATIONAL BR1_GW
For questions: www.voiceie.com
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Route List
Route List associated Route Group with members
Description BR1_INTERNATIONAL
CUCM Group SUB-PUB
Enabled Y


Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members 10.10.32.2 (All Ports)



BR1_LD
Description BR1_LD
CUCM Group SUB-PUB
Enabled Y


BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits < None >
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members 10.10.32.2 (All Ports)

HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)
For questions: www.voiceie.com
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Route List
Route List associated Route Group with members



BR1_LOCAL
Description BR1_LOCAL
CUCM Group SUB-PUB
Enabled Y


BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits < None >
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members 10.10.32.2 (All Ports)

HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 1702
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)



HQ AAR
Description HQ AAR
CUCM Group SUB-PUB
Enabled Y


HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
For questions: www.voiceie.com
v.a.l.03.09.05.cf.r05.24.06.pages
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Route List
Route List associated Route Group with members
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)



HQ_911
Description HQ_911
CUCM Group SUB-PUB
Enabled Y


HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)



HQ_BR1_TOLLBYPASS
Description HQ_BR1_TOLLBYPASS
CUCM Group SUB-PUB
Enabled Y


BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 9
Members 10.10.32.2 (All Ports)

HQ_GW
Calling Party Transformations
For questions: www.voiceie.com
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Route List
Route List associated Route Group with members
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 1702
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)



HQ_BR2
Description HQ_BR2
CUCM Group SUB-PUB
Enabled Y


BR2_TRUNK
Calling Party Transformations
Use Calling Party's External Mask Off
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members BR2 (All Ports)

HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 01144207630
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)


For questions: www.voiceie.com
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Route List
Route List associated Route Group with members

HQ_BR2_TOLLBYPASS
Description HQ_BR2_TOLLBYPASS
CUCM Group SUB-PUB
Enabled Y


BR2_TRUNK
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 9
Members BR2 (All Ports)

HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 01144207
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)



HQ_INTERNATIONAL
Description HQ_INTERNATIONAL
CUCM Group SUB-PUB
Enabled Y


HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
For questions: www.voiceie.com
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Route List
Route List associated Route Group with members
Prefix Digits (Outgoing Calls)
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)



HQ_LD
Description HQ_LD
CUCM Group SUB-PUB
Enabled Y


HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)

BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 9
Members 10.10.32.2 (All Ports)



HQ_LOCAL
Description HQ_LOCAL
CUCM Group SUB-PUB
Enabled Y

HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
For questions: www.voiceie.com
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Route List
Route List associated Route Group with members
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)

BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 91702
Members 10.10.32.2 (All Ports)


For questions: www.voiceie.com


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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:
For questions: www.voiceie.com
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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:
For questions: www.voiceie.com
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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
DNs 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 arent 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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Application User
User ID Information
CtiGw

Application User Information
Presence Group Standard Presence group
Accept Presence Subscription N
Accept Out-of-dialog REFER N
Accept Unsolicited Notification N
Accept Replaces Header N
Device Information
Controlled Devices SEP0016C72D3DCC
SEP003094C27864
SEP003094C2DA38
SEP003094C3DCF9
CAPF Information
Associated CAPF Profiles
Permissions Information
Groups Standard CTI Allow Control of All Devices
Standard CTI Enabled
Roles Standard CTI Allow Control of All Devices
Standard CTI Enabled


IPCC

Application User Information
Presence Group Standard Presence group
Accept Presence Subscription N
Accept Out-of-dialog REFER N
Accept Unsolicited Notification N
Accept Replaces Header N
Device Information
Controlled Devices
CAPF Information
Associated CAPF Profiles
Permissions Information
Groups Standard AXL User
Roles Standard AXL API Access


Unity

Application User Information
Presence Group Standard Presence group
Accept Presence Subscription N
Accept Out-of-dialog REFER N
Accept Unsolicited Notification N
Accept Replaces Header N
Device Information
Controlled Devices
CAPF Information
Associated CAPF Profiles
Permissions Information
Groups Standard AXL User
Roles Standard AXL API Access

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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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Application User
User ID Information

jtapi1_1

Application User Information
Presence Group Standard Presence group
Accept Presence Subscription N
Accept Out-of-dialog REFER N
Accept Unsolicited Notification N
Accept Replaces Header N
Device Information
Controlled Devices AA_1015
AA_1016
AA_1017
AA_1018
AA_1019
AA
ICD
CAPF Information
Associated CAPF Profiles
Permissions Information
Groups Standard CTI Enabled
Roles Standard CTI Enabled


rmjtapi1

Application User Information
Presence Group Standard Presence group
Accept Presence Subscription N
Accept Out-of-dialog REFER N
Accept Unsolicited Notification N
Accept Replaces Header N
Device Information
Controlled Devices SEP0016C72D3DCC
SEP003094C27864
SEP003094C2DA38
SEP003094C3DCF9
CAPF Information
Associated CAPF Profiles
Permissions Information
Groups Standard CTI Enabled
Roles Standard CTI Enabled

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The following End Users will also need to be created with
the correct settings from the table:



End User (Condensed)
User ID Telephone
Number
Controlled
Devices
Controlled
Profiles
Associated
Extensions
Groups Roles
BR1 2001 SEP003094C3DCF9 2001 in PT-
BR1-Internal
2001 - IPCC
Standard CCM End
Users
Standard CTI
Enabled
Standard CCM End Users
Standard CCMUSER
Administration
Standard CTI Enabled
BR2 2002 SEP0016C72D3DCC 2002 in PT-
BR1-Internal
Standard CCM End
Users
Standard CTI
Enabled
Standard CCM End Users
Standard CCMUSER
Administration
Standard CTI Enabled
HQ1 1001 SEP003094C27864
UPCHQ1
1001 in PT-
HQ-Internal
1001 - IPCC
Standard CCM End
Users
Standard CTI
Enabled
Standard CCM End Users
Standard CCMUSER
Administration
Standard CTI Enabled
HQ2 1002 SEP003094C2DA38 1002 in PT-
HQ-Internal
Standard CCM End
Users
Standard CTI
Enabled
Standard CCM End Users
Standard CCMUSER
Administration
Standard CTI Enabled
crsadmin < None >
john 1007 John 1007 in PT-
HQ-Internal
Standard CCM End
Users
Standard CTI
Enabled
Standard CCM End Users
Standard CCMUSER
Administration
Standard CTI Enabled

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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 conguration:
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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 dont see any resources in IPCC Express then you probably
havent 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 ACLs:
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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 130140
using the IOS DHCP on the BR1 Router.
5. Configure BR2 phones to have IP addresses 210220
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 pins
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 - Dont 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 - Dont 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 - Dont 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 - Dont 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 dont 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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Calling Search Space
Name Description Route Partitions for this Calling Search Space
CSS_BR1_AAR CSS_BR1_AAR PT-BR1-AAR
CSS_BR1_INT CSS_BR1_INT PT-BR1-911
PT-BR1-Internal
PT-BR1-TOLLBYPASS
PT-HQ-Internal
PT-BR2-Internal
PT-BR1-Local
PT-BR1-LD
PT-BR1-INT
CSS_BR1_INTERNAL CSS_BR1_INTERNAL PT-BR1-911
PT-BR1-Internal
PT-BR1-TOLLBYPASS
PT-HQ-Internal
PT-BR2-Internal
CSS_BR1_LD CSS_BR1_LD PT-BR1-911
PT-BR1-Internal
PT-BR1-TOLLBYPASS
PT-HQ-Internal
PT-BR2-Internal
PT-BR1-Local
PT-BR1-LD
CSS_BR1_LOCAL CSS_BR1_LOCAL PT-BR1-911
PT-BR1-Internal
PT-BR1-TOLLBYPASS
PT-HQ-Internal
PT-BR2-Internal
PT-BR1-Local
CSS_BR2 CSS_BR2 PT-BR1-Internal
PT-BR2-Internal
PT-HQ-Internal
PT-BR2-TOLLBYPASS
CSS_HQ_AAR CSS_HQ_AAR PT-HQ-AAR
CSS_HQ_INT CSS_HQ_INT PT-HQ-911
PT-HQ-Internal
PT-HQ-TOLLBYPASS
PT-BR1-Internal
PT-BR2-Internal
PT-HQ-Local
PT-HQ-LD
PT-HQ-INT
CSS_HQ_INTERNAL CSS_HQ_INTERNAL PT-HQ-911
PT-HQ-Internal
PT-HQ-TOLLBYPASS
PT-BR1-Internal
PT-BR2-Internal
CSS_HQ_LD CSS_HQ_LD PT-HQ-911
PT-HQ-Internal
PT-HQ-TOLLBYPASS
PT-BR1-Internal
PT-BR2-Internal
PT-HQ-Local
PT-HQ-LD
CSS_HQ_LOCAL CSS_HQ_LOCAL PT-HQ-911
PT-HQ-Internal
PT-HQ-TOLLBYPASS
PT-BR1-Internal
PT-BR2-Internal
PT-HQ-Local
CSS_VOICEMAIL CSS_VOICEMAIL PT-BR1-Internal
PT-HQ-Internal


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 DNs:
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 DNs:
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 DNs 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-dns.
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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Route List
Route List associated Route Group with members
BR1 AAR
Description BR1 AAR
CUCM Group SUB-PUB
Enabled Y


BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits < None >
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members 10.10.32.2 (All Ports)



BR1_911
Description BR1_911
CUCM Group SUB-PUB
Enabled Y


BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members 10.10.32.2 (All Ports)



BR1_BR2
Description BR1_BR2
CUCM Group SUB-PUB
Enabled Y


BR2_TRUNK
Calling Party Transformations
Use Calling Party's External Mask Off
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
For questions: www.voiceie.com
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Route List
Route List associated Route Group with members
Prefix Digits (Outgoing Calls)
Members BR2 (All Ports)

BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 901144207630
Members 10.10.32.2 (All Ports)



BR1_BR2_TOLLBYPASS
Description BR1_BR2_TOLLBYPASS
CUCM Group SUB-PUB
Enabled Y


BR2_TRUNK
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 9
Members BR2 (All Ports)

BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
For questions: www.voiceie.com
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Route List
Route List associated Route Group with members
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 901144207
Members 10.10.32.2 (All Ports)



BR1_HQ_TOLLBYPASS
Description BR1_HQ_TOLLBYPASS
CUCM Group SUB-PUB
Enabled Y


HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)

BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 91702
Members 10.10.32.2 (All Ports)



BR1_INTERNATIONAL BR1_GW
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Route List
Route List associated Route Group with members
Description BR1_INTERNATIONAL
CUCM Group SUB-PUB
Enabled Y


Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members 10.10.32.2 (All Ports)



BR1_LD
Description BR1_LD
CUCM Group SUB-PUB
Enabled Y


BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits < None >
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members 10.10.32.2 (All Ports)

HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)
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Route List
Route List associated Route Group with members



BR1_LOCAL
Description BR1_LOCAL
CUCM Group SUB-PUB
Enabled Y


BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits < None >
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members 10.10.32.2 (All Ports)

HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 1702
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)



HQ AAR
Description HQ AAR
CUCM Group SUB-PUB
Enabled Y


HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
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Route List
Route List associated Route Group with members
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)



HQ_911
Description HQ_911
CUCM Group SUB-PUB
Enabled Y


HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)



HQ_BR1_TOLLBYPASS
Description HQ_BR1_TOLLBYPASS
CUCM Group SUB-PUB
Enabled Y


BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 9
Members 10.10.32.2 (All Ports)

HQ_GW
Calling Party Transformations
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Route List
Route List associated Route Group with members
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 1702
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)



HQ_BR2
Description HQ_BR2
CUCM Group SUB-PUB
Enabled Y


BR2_TRUNK
Calling Party Transformations
Use Calling Party's External Mask Off
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members BR2 (All Ports)

HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 01144207630
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)


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Route List
Route List associated Route Group with members

HQ_BR2_TOLLBYPASS
Description HQ_BR2_TOLLBYPASS
CUCM Group SUB-PUB
Enabled Y


BR2_TRUNK
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 9
Members BR2 (All Ports)

HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 01144207
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)



HQ_INTERNATIONAL
Description HQ_INTERNATIONAL
CUCM Group SUB-PUB
Enabled Y


HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
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Route List
Route List associated Route Group with members
Prefix Digits (Outgoing Calls)
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)



HQ_LD
Description HQ_LD
CUCM Group SUB-PUB
Enabled Y


HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)

BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 9
Members 10.10.32.2 (All Ports)



HQ_LOCAL
Description HQ_LOCAL
CUCM Group SUB-PUB
Enabled Y

HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
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Route List
Route List associated Route Group with members
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)

BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 91702
Members 10.10.32.2 (All Ports)


For questions: www.voiceie.com


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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
DNs 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 arent 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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Application User
User ID Information
CtiGw

Application User Information
Presence Group Standard Presence group
Accept Presence Subscription N
Accept Out-of-dialog REFER N
Accept Unsolicited Notification N
Accept Replaces Header N
Device Information
Controlled Devices SEP0016C72D3DCC
SEP003094C27864
SEP003094C2DA38
SEP003094C3DCF9
CAPF Information
Associated CAPF Profiles
Permissions Information
Groups Standard CTI Allow Control of All Devices
Standard CTI Enabled
Roles Standard CTI Allow Control of All Devices
Standard CTI Enabled


IPCC

Application User Information
Presence Group Standard Presence group
Accept Presence Subscription N
Accept Out-of-dialog REFER N
Accept Unsolicited Notification N
Accept Replaces Header N
Device Information
Controlled Devices
CAPF Information
Associated CAPF Profiles
Permissions Information
Groups Standard AXL User
Roles Standard AXL API Access


Unity

Application User Information
Presence Group Standard Presence group
Accept Presence Subscription N
Accept Out-of-dialog REFER N
Accept Unsolicited Notification N
Accept Replaces Header N
Device Information
Controlled Devices
CAPF Information
Associated CAPF Profiles
Permissions Information
Groups Standard AXL User
Roles Standard AXL API Access

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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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Application User
User ID Information

jtapi1_1

Application User Information
Presence Group Standard Presence group
Accept Presence Subscription N
Accept Out-of-dialog REFER N
Accept Unsolicited Notification N
Accept Replaces Header N
Device Information
Controlled Devices AA_1015
AA_1016
AA_1017
AA_1018
AA_1019
AA
ICD
CAPF Information
Associated CAPF Profiles
Permissions Information
Groups Standard CTI Enabled
Roles Standard CTI Enabled


rmjtapi1

Application User Information
Presence Group Standard Presence group
Accept Presence Subscription N
Accept Out-of-dialog REFER N
Accept Unsolicited Notification N
Accept Replaces Header N
Device Information
Controlled Devices SEP0016C72D3DCC
SEP003094C27864
SEP003094C2DA38
SEP003094C3DCF9
CAPF Information
Associated CAPF Profiles
Permissions Information
Groups Standard CTI Enabled
Roles Standard CTI Enabled

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The following End Users will also need to be created with
the correct settings from the table:



End User (Condensed)
User ID Telephone
Number
Controlled
Devices
Controlled
Profiles
Associated
Extensions
Groups Roles
BR1 2001 SEP003094C3DCF9 2001 in PT-
BR1-Internal
2001 - IPCC
Standard CCM End
Users
Standard CTI
Enabled
Standard CCM End Users
Standard CCMUSER
Administration
Standard CTI Enabled
BR2 2002 SEP0016C72D3DCC 2002 in PT-
BR1-Internal
Standard CCM End
Users
Standard CTI
Enabled
Standard CCM End Users
Standard CCMUSER
Administration
Standard CTI Enabled
HQ1 1001 SEP003094C27864
UPCHQ1
1001 in PT-
HQ-Internal
1001 - IPCC
Standard CCM End
Users
Standard CTI
Enabled
Standard CCM End Users
Standard CCMUSER
Administration
Standard CTI Enabled
HQ2 1002 SEP003094C2DA38 1002 in PT-
HQ-Internal
Standard CCM End
Users
Standard CTI
Enabled
Standard CCM End Users
Standard CCMUSER
Administration
Standard CTI Enabled
crsadmin < None >
john 1007 John 1007 in PT-
HQ-Internal
Standard CCM End
Users
Standard CTI
Enabled
Standard CCM End Users
Standard CCMUSER
Administration
Standard CTI Enabled

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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 conguration:
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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-dns 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 dont see any resources in IPCC Express then you probably
havent 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 ACLs:
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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 130140
using the IOS DHCP on the BR1 Router.
5. Configure BR2 phones to have IP addresses 210220
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 pins
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 - Dont 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 - Dont 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 - Dont 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 - Dont 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 dont 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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Calling Search Space
Name Description Route Partitions for this Calling Search Space
CSS_BR1_AAR CSS_BR1_AAR PT-BR1-AAR
CSS_BR1_INT CSS_BR1_INT PT-BR1-911
PT-BR1-Internal
PT-BR1-TOLLBYPASS
PT-HQ-Internal
PT-BR2-Internal
PT-BR1-Local
PT-BR1-LD
PT-BR1-INT
CSS_BR1_INTERNAL CSS_BR1_INTERNAL PT-BR1-911
PT-BR1-Internal
PT-BR1-TOLLBYPASS
PT-HQ-Internal
PT-BR2-Internal
CSS_BR1_LD CSS_BR1_LD PT-BR1-911
PT-BR1-Internal
PT-BR1-TOLLBYPASS
PT-HQ-Internal
PT-BR2-Internal
PT-BR1-Local
PT-BR1-LD
CSS_BR1_LOCAL CSS_BR1_LOCAL PT-BR1-911
PT-BR1-Internal
PT-BR1-TOLLBYPASS
PT-HQ-Internal
PT-BR2-Internal
PT-BR1-Local
CSS_BR2 CSS_BR2 PT-BR1-Internal
PT-BR2-Internal
PT-HQ-Internal
PT-BR2-TOLLBYPASS
CSS_HQ_AAR CSS_HQ_AAR PT-HQ-AAR
CSS_HQ_INT CSS_HQ_INT PT-HQ-911
PT-HQ-Internal
PT-HQ-TOLLBYPASS
PT-BR1-Internal
PT-BR2-Internal
PT-HQ-Local
PT-HQ-LD
PT-HQ-INT
CSS_HQ_INTERNAL CSS_HQ_INTERNAL PT-HQ-911
PT-HQ-Internal
PT-HQ-TOLLBYPASS
PT-BR1-Internal
PT-BR2-Internal
CSS_HQ_LD CSS_HQ_LD PT-HQ-911
PT-HQ-Internal
PT-HQ-TOLLBYPASS
PT-BR1-Internal
PT-BR2-Internal
PT-HQ-Local
PT-HQ-LD
CSS_HQ_LOCAL CSS_HQ_LOCAL PT-HQ-911
PT-HQ-Internal
PT-HQ-TOLLBYPASS
PT-BR1-Internal
PT-BR2-Internal
PT-HQ-Local
CSS_VOICEMAIL CSS_VOICEMAIL PT-BR1-Internal
PT-HQ-Internal


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 DNs:
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 DNs:
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 DNs 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-dns.
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
For questions: www.voiceie.com
v.a.l.03.09.05.cf.r05.24.06.pages
618
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Copyright 2009, Network Learning, Incorporated
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
For questions: www.voiceie.com
v.a.l.03.09.05.cf.r05.24.06.pages
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Copyright 2009, Network Learning, Incorporated
!
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.
For questions: www.voiceie.com
v.a.l.03.09.05.cf.r05.24.06.pages
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Copyright 2009, Network Learning, Incorporated
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.
For questions: www.voiceie.com
v.a.l.03.09.05.cf.r05.24.06.pages
621
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Copyright 2009, Network Learning, Incorporated
Next we will need to configure the gateway in
CallManager:
For questions: www.voiceie.com
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For questions: www.voiceie.com
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623
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Copyright 2009, Network Learning, Incorporated
For questions: www.voiceie.com
v.a.l.03.09.05.cf.r05.24.06.pages
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Copyright 2009, Network Learning, Incorporated
!
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.
For questions: www.voiceie.com
v.a.l.03.09.05.cf.r05.24.06.pages
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Copyright 2009, Network Learning, Incorporated
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
For questions: www.voiceie.com
v.a.l.03.09.05.cf.r05.24.06.pages
626
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Copyright 2009, Network Learning, Incorporated
Next we will need to setup BR1 as an H.323 gateway in
CallManager:
For questions: www.voiceie.com
v.a.l.03.09.05.cf.r05.24.06.pages
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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!
For questions: www.voiceie.com
v.a.l.03.09.05.cf.r05.24.06.pages
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Copyright 2009, Network Learning, Incorporated
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
For questions: www.voiceie.com
v.a.l.03.09.05.cf.r05.24.06.pages
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Copyright 2009, Network Learning, Incorporated
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.
For questions: www.voiceie.com
v.a.l.03.09.05.cf.r05.24.06.pages
630
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Copyright 2009, Network Learning, Incorporated
Now we need to add the trunk:
For questions: www.voiceie.com
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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#
For questions: www.voiceie.com
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For questions: www.voiceie.com
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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.
For questions: www.voiceie.com
v.a.l.03.09.05.cf.r05.24.06.pages
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Copyright 2009, Network Learning, Incorporated
For questions: www.voiceie.com
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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
For questions: www.voiceie.com
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Copyright 2009, Network Learning, Incorporated
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:
For questions: www.voiceie.com
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For questions: www.voiceie.com
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Then next the route lists in the following tables with
the proper settings:
For questions: www.voiceie.com
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Route List
Route List associated Route Group with members
BR1 AAR
Description BR1 AAR
CUCM Group SUB-PUB
Enabled Y


BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits < None >
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members 10.10.32.2 (All Ports)



BR1_911
Description BR1_911
CUCM Group SUB-PUB
Enabled Y


BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members 10.10.32.2 (All Ports)



BR1_BR2
Description BR1_BR2
CUCM Group SUB-PUB
Enabled Y


BR2_TRUNK
Calling Party Transformations
Use Calling Party's External Mask Off
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
For questions: www.voiceie.com
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Route List
Route List associated Route Group with members
Prefix Digits (Outgoing Calls)
Members BR2 (All Ports)

BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 901144207630
Members 10.10.32.2 (All Ports)



BR1_BR2_TOLLBYPASS
Description BR1_BR2_TOLLBYPASS
CUCM Group SUB-PUB
Enabled Y


BR2_TRUNK
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 9
Members BR2 (All Ports)

BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
For questions: www.voiceie.com
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Route List
Route List associated Route Group with members
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 901144207
Members 10.10.32.2 (All Ports)



BR1_HQ_TOLLBYPASS
Description BR1_HQ_TOLLBYPASS
CUCM Group SUB-PUB
Enabled Y


HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)

BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 91702
Members 10.10.32.2 (All Ports)



BR1_INTERNATIONAL BR1_GW
For questions: www.voiceie.com
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Route List
Route List associated Route Group with members
Description BR1_INTERNATIONAL
CUCM Group SUB-PUB
Enabled Y


Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members 10.10.32.2 (All Ports)



BR1_LD
Description BR1_LD
CUCM Group SUB-PUB
Enabled Y


BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits < None >
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members 10.10.32.2 (All Ports)

HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)
For questions: www.voiceie.com
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Route List
Route List associated Route Group with members



BR1_LOCAL
Description BR1_LOCAL
CUCM Group SUB-PUB
Enabled Y


BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits < None >
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members 10.10.32.2 (All Ports)

HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 1702
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)



HQ AAR
Description HQ AAR
CUCM Group SUB-PUB
Enabled Y


HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
For questions: www.voiceie.com
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Route List
Route List associated Route Group with members
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)



HQ_911
Description HQ_911
CUCM Group SUB-PUB
Enabled Y


HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)



HQ_BR1_TOLLBYPASS
Description HQ_BR1_TOLLBYPASS
CUCM Group SUB-PUB
Enabled Y


BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 9
Members 10.10.32.2 (All Ports)

HQ_GW
Calling Party Transformations
For questions: www.voiceie.com
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Route List
Route List associated Route Group with members
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 1702
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)



HQ_BR2
Description HQ_BR2
CUCM Group SUB-PUB
Enabled Y


BR2_TRUNK
Calling Party Transformations
Use Calling Party's External Mask Off
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members BR2 (All Ports)

HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 01144207630
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)


For questions: www.voiceie.com
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Copyright 2009, Network Learning, Incorporated


Route List
Route List associated Route Group with members

HQ_BR2_TOLLBYPASS
Description HQ_BR2_TOLLBYPASS
CUCM Group SUB-PUB
Enabled Y


BR2_TRUNK
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 9
Members BR2 (All Ports)

HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 01144207
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)



HQ_INTERNATIONAL
Description HQ_INTERNATIONAL
CUCM Group SUB-PUB
Enabled Y


HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
For questions: www.voiceie.com
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Copyright 2009, Network Learning, Incorporated


Route List
Route List associated Route Group with members
Prefix Digits (Outgoing Calls)
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)



HQ_LD
Description HQ_LD
CUCM Group SUB-PUB
Enabled Y


HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)

BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 9
Members 10.10.32.2 (All Ports)



HQ_LOCAL
Description HQ_LOCAL
CUCM Group SUB-PUB
Enabled Y

HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
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Route List
Route List associated Route Group with members
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)

BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 91702
Members 10.10.32.2 (All Ports)


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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. Remember the CME will
automatically register its DNs 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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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 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
DNs 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 arent 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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Application User
User ID Information
CtiGw

Application User Information
Presence Group Standard Presence group
Accept Presence Subscription N
Accept Out-of-dialog REFER N
Accept Unsolicited Notification N
Accept Replaces Header N
Device Information
Controlled Devices SEP0016C72D3DCC
SEP003094C27864
SEP003094C2DA38
SEP003094C3DCF9
CAPF Information
Associated CAPF Profiles
Permissions Information
Groups Standard CTI Allow Control of All Devices
Standard CTI Enabled
Roles Standard CTI Allow Control of All Devices
Standard CTI Enabled


IPCC

Application User Information
Presence Group Standard Presence group
Accept Presence Subscription N
Accept Out-of-dialog REFER N
Accept Unsolicited Notification N
Accept Replaces Header N
Device Information
Controlled Devices
CAPF Information
Associated CAPF Profiles
Permissions Information
Groups Standard AXL User
Roles Standard AXL API Access


Unity

Application User Information
Presence Group Standard Presence group
Accept Presence Subscription N
Accept Out-of-dialog REFER N
Accept Unsolicited Notification N
Accept Replaces Header N
Device Information
Controlled Devices
CAPF Information
Associated CAPF Profiles
Permissions Information
Groups Standard AXL User
Roles Standard AXL API Access

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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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Application User
User ID Information

jtapi1_1

Application User Information
Presence Group Standard Presence group
Accept Presence Subscription N
Accept Out-of-dialog REFER N
Accept Unsolicited Notification N
Accept Replaces Header N
Device Information
Controlled Devices AA_1015
AA_1016
AA_1017
AA_1018
AA_1019
AA
ICD
CAPF Information
Associated CAPF Profiles
Permissions Information
Groups Standard CTI Enabled
Roles Standard CTI Enabled


rmjtapi1

Application User Information
Presence Group Standard Presence group
Accept Presence Subscription N
Accept Out-of-dialog REFER N
Accept Unsolicited Notification N
Accept Replaces Header N
Device Information
Controlled Devices SEP0016C72D3DCC
SEP003094C27864
SEP003094C2DA38
SEP003094C3DCF9
CAPF Information
Associated CAPF Profiles
Permissions Information
Groups Standard CTI Enabled
Roles Standard CTI Enabled

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The following End Users will also need to be created with
the correct settings from the table:



End User (Condensed)
User ID Telephone
Number
Controlled
Devices
Controlled
Profiles
Associated
Extensions
Groups Roles
BR1 2001 SEP003094C3DCF9 2001 in PT-
BR1-Internal
2001 - IPCC
Standard CCM End
Users
Standard CTI
Enabled
Standard CCM End Users
Standard CCMUSER
Administration
Standard CTI Enabled
BR2 2002 SEP0016C72D3DCC 2002 in PT-
BR1-Internal
Standard CCM End
Users
Standard CTI
Enabled
Standard CCM End Users
Standard CCMUSER
Administration
Standard CTI Enabled
HQ1 1001 SEP003094C27864
UPCHQ1
1001 in PT-
HQ-Internal
1001 - IPCC
Standard CCM End
Users
Standard CTI
Enabled
Standard CCM End Users
Standard CCMUSER
Administration
Standard CTI Enabled
HQ2 1002 SEP003094C2DA38 1002 in PT-
HQ-Internal
Standard CCM End
Users
Standard CTI
Enabled
Standard CCM End Users
Standard CCMUSER
Administration
Standard CTI Enabled
crsadmin < None >
john 1007 John 1007 in PT-
HQ-Internal
Standard CCM End
Users
Standard CTI
Enabled
Standard CCM End Users
Standard CCMUSER
Administration
Standard CTI Enabled

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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 conguration:
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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-dns 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 dont see any resources in IPCC Express then you probably
havent 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 ACLs:
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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 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 130140
using the IOS DHCP on the BR1 Router.
5. Configure BR2 phones to have IP addresses 210220
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.
For questions: www.voiceie.com
v.a.l.03.09.05.cf.r05.24.06.pages
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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
For questions: www.voiceie.com
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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
For questions: www.voiceie.com
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Copyright 2009, Network Learning, Incorporated
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.
For questions: www.voiceie.com
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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.
For questions: www.voiceie.com
v.a.l.03.09.05.cf.r05.24.06.pages
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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.
For questions: www.voiceie.com
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Copyright 2009, Network Learning, Incorporated
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.
For questions: www.voiceie.com
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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.
For questions: www.voiceie.com
v.a.l.03.09.05.cf.r05.24.06.pages
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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 pins
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.
For questions: www.voiceie.com
v.a.l.03.09.05.cf.r05.24.06.pages
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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.
For questions: www.voiceie.com
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2. Make sure HQ Phone 1 can see the voicemail in their
voicemail box through the CUPC client.
For questions: www.voiceie.com
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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
For questions: www.voiceie.com
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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 - Dont 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.
For questions: www.voiceie.com
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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 - Dont 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
For questions: www.voiceie.com
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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 - Dont 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
For questions: www.voiceie.com
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For questions: www.voiceie.com
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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
For questions: www.voiceie.com
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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
For questions: www.voiceie.com
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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 - Dont forget to set the time zone settings in CallManager as it may not be
mentioned in the phone setup tasks.
For questions: www.voiceie.com
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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 dont 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.
For questions: www.voiceie.com
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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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Calling Search Space
Name Description Route Partitions for this Calling Search Space
CSS_BR1_AAR CSS_BR1_AAR PT-BR1-AAR
CSS_BR1_INT CSS_BR1_INT PT-BR1-911
PT-BR1-Internal
PT-BR1-TOLLBYPASS
PT-HQ-Internal
PT-BR2-Internal
PT-BR1-Local
PT-BR1-LD
PT-BR1-INT
CSS_BR1_INTERNAL CSS_BR1_INTERNAL PT-BR1-911
PT-BR1-Internal
PT-BR1-TOLLBYPASS
PT-HQ-Internal
PT-BR2-Internal
CSS_BR1_LD CSS_BR1_LD PT-BR1-911
PT-BR1-Internal
PT-BR1-TOLLBYPASS
PT-HQ-Internal
PT-BR2-Internal
PT-BR1-Local
PT-BR1-LD
CSS_BR1_LOCAL CSS_BR1_LOCAL PT-BR1-911
PT-BR1-Internal
PT-BR1-TOLLBYPASS
PT-HQ-Internal
PT-BR2-Internal
PT-BR1-Local
CSS_BR2 CSS_BR2 PT-BR1-Internal
PT-BR2-Internal
PT-HQ-Internal
PT-BR2-TOLLBYPASS
CSS_HQ_AAR CSS_HQ_AAR PT-HQ-AAR
CSS_HQ_INT CSS_HQ_INT PT-HQ-911
PT-HQ-Internal
PT-HQ-TOLLBYPASS
PT-BR1-Internal
PT-BR2-Internal
PT-HQ-Local
PT-HQ-LD
PT-HQ-INT
CSS_HQ_INTERNAL CSS_HQ_INTERNAL PT-HQ-911
PT-HQ-Internal
PT-HQ-TOLLBYPASS
PT-BR1-Internal
PT-BR2-Internal
CSS_HQ_LD CSS_HQ_LD PT-HQ-911
PT-HQ-Internal
PT-HQ-TOLLBYPASS
PT-BR1-Internal
PT-BR2-Internal
PT-HQ-Local
PT-HQ-LD
CSS_HQ_LOCAL CSS_HQ_LOCAL PT-HQ-911
PT-HQ-Internal
PT-HQ-TOLLBYPASS
PT-BR1-Internal
PT-BR2-Internal
PT-HQ-Local
CSS_VOICEMAIL CSS_VOICEMAIL PT-BR1-Internal
PT-HQ-Internal


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 DNs:
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 DNs:
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 DNs 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-dns.
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.
For questions: www.voiceie.com
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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
For questions: www.voiceie.com
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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.
For questions: www.voiceie.com
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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#
For questions: www.voiceie.com
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For questions: www.voiceie.com
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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.
For questions: www.voiceie.com
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For questions: www.voiceie.com
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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
For questions: www.voiceie.com
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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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Route List
Route List associated Route Group with members
BR1 AAR
Description BR1 AAR
CUCM Group SUB-PUB
Enabled Y


BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits < None >
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members 10.10.32.2 (All Ports)



BR1_911
Description BR1_911
CUCM Group SUB-PUB
Enabled Y


BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members 10.10.32.2 (All Ports)



BR1_BR2
Description BR1_BR2
CUCM Group SUB-PUB
Enabled Y


BR2_TRUNK
Calling Party Transformations
Use Calling Party's External Mask Off
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
For questions: www.voiceie.com
v.a.l.03.09.05.cf.r05.24.06.pages
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Route List
Route List associated Route Group with members
Prefix Digits (Outgoing Calls)
Members BR2 (All Ports)

BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 901144207630
Members 10.10.32.2 (All Ports)



BR1_BR2_TOLLBYPASS
Description BR1_BR2_TOLLBYPASS
CUCM Group SUB-PUB
Enabled Y


BR2_TRUNK
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 9
Members BR2 (All Ports)

BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
For questions: www.voiceie.com
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Route List
Route List associated Route Group with members
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 901144207
Members 10.10.32.2 (All Ports)



BR1_HQ_TOLLBYPASS
Description BR1_HQ_TOLLBYPASS
CUCM Group SUB-PUB
Enabled Y


HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)

BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 91702
Members 10.10.32.2 (All Ports)



BR1_INTERNATIONAL BR1_GW
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Route List
Route List associated Route Group with members
Description BR1_INTERNATIONAL
CUCM Group SUB-PUB
Enabled Y


Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members 10.10.32.2 (All Ports)



BR1_LD
Description BR1_LD
CUCM Group SUB-PUB
Enabled Y


BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits < None >
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members 10.10.32.2 (All Ports)

HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)
For questions: www.voiceie.com
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Route List
Route List associated Route Group with members



BR1_LOCAL
Description BR1_LOCAL
CUCM Group SUB-PUB
Enabled Y


BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits < None >
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members 10.10.32.2 (All Ports)

HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 1702
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)



HQ AAR
Description HQ AAR
CUCM Group SUB-PUB
Enabled Y


HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
For questions: www.voiceie.com
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Route List
Route List associated Route Group with members
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)



HQ_911
Description HQ_911
CUCM Group SUB-PUB
Enabled Y


HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)



HQ_BR1_TOLLBYPASS
Description HQ_BR1_TOLLBYPASS
CUCM Group SUB-PUB
Enabled Y


BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 9
Members 10.10.32.2 (All Ports)

HQ_GW
Calling Party Transformations
For questions: www.voiceie.com
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Route List
Route List associated Route Group with members
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 1702
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)



HQ_BR2
Description HQ_BR2
CUCM Group SUB-PUB
Enabled Y


BR2_TRUNK
Calling Party Transformations
Use Calling Party's External Mask Off
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members BR2 (All Ports)

HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 01144207630
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)


For questions: www.voiceie.com
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Route List
Route List associated Route Group with members

HQ_BR2_TOLLBYPASS
Description HQ_BR2_TOLLBYPASS
CUCM Group SUB-PUB
Enabled Y


BR2_TRUNK
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 9
Members BR2 (All Ports)

HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 01144207
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)



HQ_INTERNATIONAL
Description HQ_INTERNATIONAL
CUCM Group SUB-PUB
Enabled Y


HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
For questions: www.voiceie.com
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Route List
Route List associated Route Group with members
Prefix Digits (Outgoing Calls)
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)



HQ_LD
Description HQ_LD
CUCM Group SUB-PUB
Enabled Y


HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)

BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 9
Members 10.10.32.2 (All Ports)



HQ_LOCAL
Description HQ_LOCAL
CUCM Group SUB-PUB
Enabled Y

HQ_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
For questions: www.voiceie.com
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Route List
Route List associated Route Group with members
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls)
Members S0/SU0/DS1-0@HQ_Rtr (All Ports)

BR1_GW
Calling Party Transformations
Use Calling Party's External Mask On
Calling Party Transform Mask
Prefix Digits (Outgoing Calls)
Called Party Transformations
Discard Digits PreDot
Called Party Transform Mask
Prefix Digits (Outgoing Calls) 91702
Members 10.10.32.2 (All Ports)


For questions: www.voiceie.com


v.a.l.03.09.05.cf.r05.24.06.pages
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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:
For questions: www.voiceie.com
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For questions: www.voiceie.com
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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
For questions: www.voiceie.com
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For questions: www.voiceie.com
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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
For questions: www.voiceie.com
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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
For questions: www.voiceie.com
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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
For questions: www.voiceie.com
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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:
For questions: www.voiceie.com
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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
For questions: www.voiceie.com
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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.
For questions: www.voiceie.com
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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
DNs 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 arent 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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Application User
User ID Information
CtiGw

Application User Information
Presence Group Standard Presence group
Accept Presence Subscription N
Accept Out-of-dialog REFER N
Accept Unsolicited Notification N
Accept Replaces Header N
Device Information
Controlled Devices SEP0016C72D3DCC
SEP003094C27864
SEP003094C2DA38
SEP003094C3DCF9
CAPF Information
Associated CAPF Profiles
Permissions Information
Groups Standard CTI Allow Control of All Devices
Standard CTI Enabled
Roles Standard CTI Allow Control of All Devices
Standard CTI Enabled


IPCC

Application User Information
Presence Group Standard Presence group
Accept Presence Subscription N
Accept Out-of-dialog REFER N
Accept Unsolicited Notification N
Accept Replaces Header N
Device Information
Controlled Devices
CAPF Information
Associated CAPF Profiles
Permissions Information
Groups Standard AXL User
Roles Standard AXL API Access


Unity

Application User Information
Presence Group Standard Presence group
Accept Presence Subscription N
Accept Out-of-dialog REFER N
Accept Unsolicited Notification N
Accept Replaces Header N
Device Information
Controlled Devices
CAPF Information
Associated CAPF Profiles
Permissions Information
Groups Standard AXL User
Roles Standard AXL API Access

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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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Application User
User ID Information

jtapi1_1

Application User Information
Presence Group Standard Presence group
Accept Presence Subscription N
Accept Out-of-dialog REFER N
Accept Unsolicited Notification N
Accept Replaces Header N
Device Information
Controlled Devices AA_1015
AA_1016
AA_1017
AA_1018
AA_1019
AA
ICD
CAPF Information
Associated CAPF Profiles
Permissions Information
Groups Standard CTI Enabled
Roles Standard CTI Enabled


rmjtapi1

Application User Information
Presence Group Standard Presence group
Accept Presence Subscription N
Accept Out-of-dialog REFER N
Accept Unsolicited Notification N
Accept Replaces Header N
Device Information
Controlled Devices SEP0016C72D3DCC
SEP003094C27864
SEP003094C2DA38
SEP003094C3DCF9
CAPF Information
Associated CAPF Profiles
Permissions Information
Groups Standard CTI Enabled
Roles Standard CTI Enabled

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The following End Users will also need to be created with
the correct settings from the table:



End User (Condensed)
User ID Telephone
Number
Controlled
Devices
Controlled
Profiles
Associated
Extensions
Groups Roles
BR1 2001 SEP003094C3DCF9 2001 in PT-
BR1-Internal
2001 - IPCC
Standard CCM End
Users
Standard CTI
Enabled
Standard CCM End Users
Standard CCMUSER
Administration
Standard CTI Enabled
BR2 2002 SEP0016C72D3DCC 2002 in PT-
BR1-Internal
Standard CCM End
Users
Standard CTI
Enabled
Standard CCM End Users
Standard CCMUSER
Administration
Standard CTI Enabled
HQ1 1001 SEP003094C27864
UPCHQ1
1001 in PT-
HQ-Internal
1001 - IPCC
Standard CCM End
Users
Standard CTI
Enabled
Standard CCM End Users
Standard CCMUSER
Administration
Standard CTI Enabled
HQ2 1002 SEP003094C2DA38 1002 in PT-
HQ-Internal
Standard CCM End
Users
Standard CTI
Enabled
Standard CCM End Users
Standard CCMUSER
Administration
Standard CTI Enabled
crsadmin < None >
john 1007 John 1007 in PT-
HQ-Internal
Standard CCM End
Users
Standard CTI
Enabled
Standard CCM End Users
Standard CCMUSER
Administration
Standard CTI Enabled

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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 conguration:
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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-dns 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!
For questions: www.voiceie.com
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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.
For questions: www.voiceie.com
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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 dont see any resources in IPCC Express then you probably
havent 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.
For questions: www.voiceie.com
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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.
For questions: www.voiceie.com
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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:
For questions: www.voiceie.com
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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:
For questions: www.voiceie.com
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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 ACLs:
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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:
For questions: www.voiceie.com
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We can now set the proper profiles for the HQ phone 1
user:
For questions: www.voiceie.com
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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.
For questions: www.voiceie.com
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Copyright 2009, Network Learning, Incorporated
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:
For questions: www.voiceie.com
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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.


For questions: www.voiceie.com
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For questions: www.voiceie.com
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