Site VLAN VLAN ID IP Address

HQ Server 100 142.100.64.0/24

Voice 102 142.102.64.0/24
Data 202 142.202.64.0/24
SiteB Voice 302 142.102.65.0/24
Data 402 142.202.65.0/24
SiteC Voice 502 142.102.66.0/24
Data 602 142.202.66.0/24

*All Sub-interfaces and L3 VLAN interfaces for server, voice and data are assigned
IP address .254 from the respective subnets.





Server IP Address
CUCM Pub 142.100.64.11
CUCM Sub 142.100.64.12
Cisco Unity Connection 142.100.64.13
UCCX 142.100.64.14
CUPS 142.100.64.15
CUPC Test machine 142.100.64.16


Loopback IP Addresses
Site Interface IP Address
HQ Loopback0 142.1.64.254/32
SiteB Loopback0 142.1.65.254/32
SiteC Loopback0 142.1.66.254/32


Phone PSTN Numbering Plan
HQ Phone 1 1 408 202 2001
HQ Phone 2 1 408 202 2002
HQ Phone 3 1 408 202 2003
SiteB Phone 1 1 972 303 3001
SiteB Phone 2 1 972 303 3002
SiteC Phone 1 852 2404 4001
SiteC Phone 2 852 2404 4001

Phone Internal Numbering Plan
HQ Phone 1 2001
HQ Phone 2 2002
HQ Phone 3 2003
SiteB Phone 1 3001
SiteB Phone 2 3002
SiteC Phone 1 4001
SiteC Phone 2 4002



Port Assignments
R1
Gig 0/0 SW1 1/0/1
Serial 0/1/0.101 R2 Serial 0/2/0.101
Serial 0/1/0.201 R3 Serial 0/2/0.201

SW1
Fa 1/0/1 R1 Gig 0/0
Fa 1/0/3 CUCM PUB, SUB, Unity Connection
Fa 1/0/4 UCCX, CUPS
Fa 1/0/13 HQ Phone 1
Fa 1/0/14 HQ Phone 2
Fa 1/0/15 HQ Phone 3
Fa 1/0/16 CUPC Test machine

R2
Serial 0/2/0.201 R1 Serial 0/1/0.101
Fa 0/1/0 R2 Phone 1
Fa 0/1/1 R2 Phone 2

R3
Serial 0/2/0.201 Serial 0/1/0.201
Fa 0/1/0 R3 Phone 1
Fa 0/1/1 R3 Phone 2



*R2 and R3 routers have HWIC-4ESW for connecting IP Phones.
Enable password for routers and switches cisco
Username for servers administrator
Password for servers - ccievoice

Section 1: Core Knowledge Questions
There are 4 open ended questions in this section. You should answer
at least 3 out of 4 correctly to get 100% in this section. If you fail to
do so, your score will be 0% in this section.
(21 Points)

See another document for open ended















Section 2: Basic Campus Design

2.1 Voice and Data VLANs
Configure Voice VLANs for switch ports connecting to IP Phones at HQ, SiteB and
SiteC. Voice VLAN IDs for HQ, SiteB and SiteC are 102, 302 and 502 respectively.
There is a machine connected to each switch port. Configure switch ports such
that machine will be placed in an appropriate data VLAN. Data VLAN IDs for HQ,
SiteB and SiteC are 202, 402 and 602 respectively.
Refer to port assignment and VLAN Detail tables for more information.
(2 points)
2.2 DHCP Service
Configure CUCM Publisher as DHCP server to provide IP Addresses for IP Phones
at HQ and SiteB from their respective Voice subnets.
For HQ, use IP address range from 142.102.64.10/24 to 142.102.64.30/24
For SiteB, use IP address range from 142.102.65.10/24 to 142.102.65.30/24
Configure local Cisco 2811 router as DHCP server to provide IP addresses for SiteC
IP Phones from local Voice subnet.
Use IP address range from 142.102.66.10/24 to 142.102.66.30/24
(2 points)
2.3 NTP
Synchronize HQ router with external NTP source at 157.26.1.100. This External
NTP server is in UTC time zone. Configure HQ router in PST time zone which is 8
hours behind UTC.
Synchronize CUCM Publisher with loopback interface of HQ router. SiteB is in CST
time zone which is 2 hours ahead of PST.
SiteC is in Hong Kong time zone which is 8 hours ahead of UTC.
Configure CUCM such that IP phones display appropriate time according to the
time zone to which they belong. (2 points)



Section 3: Cisco Unified Communication Manager

3.1 CUCM IP Phones registration
Register IP phones at HQ, SiteB and SiteC to CUCM and assign extension numbers
as specified in the above table.
Extension-to-extension calling should use 4-digit dialing and should also deliver
calling name. You can use any trivial names such as hq ph1, siteb ph1 etc.
IP Phones should display globalized dialing number at the right hand corner e.g-
HQ Phone 1 should display +14022022001, SiteC Phone 1 should display
+85224044001.
(3 points)

3.2 IP Phone customization (Part I)
HQ phone 2 user is complaining about call transfer behavior of IP Phones. He/She
has to press Transfer softkey , dial the desired number and again press
Transfer softkey for transferring call to any number. Configure CUCM to meet
following requirements,
1) Assign Transfer key to button 3 of HQ IP Phone 2 instead of softkey.
2) While any call is active, all of the users can press Transfer softkey, dial
the desired number and once other end callback is heard they can simply
put the receiver back to the cradle of the IP phone to transfer the call.
(3 points)

3.3 IP Phone customization (Part II)
Configure directory number 2103 on line 2 of HQ Phone 3. After configuring line 2
extension, user is complaining that when calls are active on both the lines, he/she
can not take both the calls in conference using Join softkey.
Configure this feature only for HQ Phone 3 wherein User has to specifically use
join softkey to connect active calls on both the lines into single conference call.
(2 points)



Section 4: Voice Gateways and Signaling

4.1 HQ IOS MGCP T1-PRI gateway
Configure CUCM to register HQ Router controller T1 0/0/0 as IOS MGCP T1 PRI
gateway. Make sure that all inbound and outbound MGCP traffic is sourced from
the local interface 142.102.64.254/24.
Telco is sending 10-digits Direct-Inward-Dial (DID) for inbound PSTN calls. Test
the inbound calls to HQ IP Phones 408202xxxx where xxxx is extension range of
HQ IP Phones.
Verify the gateway functionality by making outgoing calls to 911 emergency
number. Calls made to this number should display 10-digit caller ID as
408202xxxx.
There is no need to test 9911 calling.
(2 points)

4.2 SiteB IOS MGCP T1-PRI gateway
Configure CUCM to register SiteB Router controller T1 0/0/0 as IOS MGCP T1 PRI
gateway. Make sure that all inbound and outbound MGCP traffic is sourced from
the local interface 142.102.65.254/24.
Telco is sending 10-digits Direct-Inward-Dial (DID) for inbound PSTN calls. Test
the inbound calls to SiteB IP Phones 972303xxxx where xxxx is extension range of
SiteB IP Phones.
Verify the gateway functionality by making outgoing calls to 911 emergency
number. Calls made to this number should display 10-digit caller ID as
972303xxxx.
There is no need to test 9911 calling.
(2 points)






4.3 SiteC IOS H323 gateway
Configure SiteC router as H323 gateway and register the same to CUCM. Use only
12 channels of E1 PRI.
Make sure that all inbound and outbound H323 traffic is sourced from the local
interface 142.102.66.254/24.
Telco is sending 8-digits Direct-Inward-Dial (DID) for inbound PSTN calls. Test the
inbound calls to SiteC IP Phones 2404xxxx where xxxx is extension range of SiteC
IP Phones.
Verify the gateway functionality by making outgoing calls to 999 emergency
number. Calls made to this number should display 8-digit caller ID as 2404xxxx.
(2 points)

Section 5: CUCM Call Routing
PSTN access code for all IP phones 9
Country code for US 1
Country code for Hong Kong - 852
National code for HQ and SiteB IP phones 1
International code for HQ and SiteB IP Phones 011
International code for SiteC IP Phones 00








5.1 CUCM Call Routing HQ Gateway
HQ PSTN provider specifications are as follows,
1) HQ PSTN provider expects proper information in called party number and
called party number type fields.
2) Called party number and called party number type information must be
set in ISDN setup messages. (Subscriber for local, National for long distance
and International for International calls).
3) You MUST not use leading digit information to signal national (1) or
international (011) calls.
4) If HQ Phone 1 makes international call to SiteC Phone 1 901185224044001,
service provider expects 85224044001 in called party number field and
International in called party number type field to route this call
properly.
5) Unknown Called party number type field is only accepted for 911
emergency calls.
By considering the above specifications, configure following requirements,
1) All HQ IP phones can make local PSTN calls by dialing 9 followed by 7 digit
PSTN number. Second digit after the access code can be anything between 2
to 9. Rest of the digits can be anything between 0 to 9. For such local calls,
PSTN should send 7-digit calling number 202xxxx along with calling name.
Also, called party number type should be set to subscriber for these calls.
Only HQ gateway should be selected and no redundancy is required.
2) All HQ IP phones can make International calls by dialing 9 followed by 011
then country code and variable length dialing digits. Calling number for such
calls should be US country code leading + i.e. - +1408202xxxx.
International calls should use only HQ gateway and no redundancy is
required.
Also, called party number type should be set to international for these
calls.
3) Configure local route group for both the type of calls mentioned above so
that it uses only HQ gateway for call routing.
(3 points)



5.1 CUCM Call Routing SiteB Gateway
SiteB PSTN provider specifications are as follows,
1) HQ PSTN provider uses leading digits in the called number to signal non-
local calls. 1 for national and 011 for international calls.
2) Called party number type information can be ignored except local calls
for which provider expects subscriber as Called party number type field.
3) If SiteB Phone 1 makes international call to SiteC Phone 1
901185224044001, service provider expects 01185224044001 in called
party number field and to route this call properly.
4) Unknown Called party number type field is only accepted for 911
emergency calls.
By considering the above specifications, configure following requirements,
1) All SiteB IP phones can make local PSTN calls by dialing 9 followed by 7
digit PSTN number. For such local calls, PSTN should send 7-digit calling
number 404xxxx along with calling name. Only SiteB gateway should be
selected and no redundancy is required.
2) If SiteB IP Phone makes national call to numbers in 408 area code, HQ
gateway should be selected to route these calls. 10-digit Calling number
1972303xxxx should be sent out to PSTN along with calling name.
3) For above calls, if HQ gateway is not reachable, it should use SiteB local
gateway. 10-digit Calling number 1972303xxxx should be sent out to PSTN
along with calling name.

(3 points)

5.3 CUCM Call Routing SiteC Gateway
SiteC PSTN provider specifications are as follows,
1) SiteC PSTN provider expects proper information in called party number
and called party number type fields.
2) Called party number and called party number type information must be
set in ISDN setup messages. (Subscriber for local, National for long distance
and International for International calls).


3) If SiteC Phone 1 makes international call to HQ Phone 1 90014082022001,
service provider expects 14082022001 in called party number field and
International in called party number type field to route this call
properly.
4) Unknown Called party number type field is only accepted for 911
emergency calls.
By considering the above specifications, configure following requirements,
1) All SiteC IP phones can make local PSTN calls by dialing 9 followed by 8-
digit PSTN number. For such local calls, PSTN should send 8-digit calling number
2404xxxx along with calling name. Also, called party number type should be set
to subscriber for these calls. Only SiteC gateway should be selected and no
redundancy is required.
2) All SiteC IP phones can make International calls by dialing 9 followed by 00
then country code and variable length dialing digits. Calling number for such calls
should be Hong kong country code leading + i.e. - +18522404xxxx.
International calls should use only SiteC gateway and no redundancy is required.
Also, called party number type should be set to international for these calls.
3) Configure local route group for both the type of calls mentioned above so
that it uses only SiteC gateway for call routing.
(4 points)

5.4 CUCM Call Routing + dialing consideration
Configure CUCM to deliver globalized dialing pattern for HQ IP phones. Use debug
isdn q931 output to verify number type information for calling and called number
sent by PSTN.
Refer to below example,
1) Make inbound call to HQ IP Phone 1 2022001 from HQ PSTN phone
2722222.
2) On HQ IP phone 1, it displays 7 digit calling number 2722222 along with
calling name as hq pstn. Do not answer this call.
3) Press directories button to go to missed call menu. After selecting missed
calls menu, this call should display globalized calling number
+14082722222.


4) Select this call from list and click dial button to call this number. This should
select HQ gateway for call routing.
5) Once call is connected, it should show To 2722222 on HQ IP phone 1
display and From 2022001 on PSTN phone display.
(3 points)

5.4 CUCM Call Routing + dialing consideration with TEHO
Configure CUCM to deliver globalized dialing pattern for HQ IP phones. Use debug
isdn q931 output to verify number type information for calling and called number
sent by PSTN.
Refer to below example,
1) Make international inbound call to HQ IP Phone 1 90014082022001 from
SiteC PSTN phone 27685555.
2) On HQ IP phone 1, it displays calling number as 85227685555 along with
calling name as sitec pstn. Do not answer this call.
3) Press directories button to go to missed call menu. After selecting missed
calls menu, this call should display globalized calling number
+85227685555.
4) Select this call from list and click dial button to call this number. This should
select SiteC gateway for call routing.
5) Once call is connected, it should show To 27685555 on HQ IP phone 1
display and From 14082022001 on PSTN phone display.
6) If SiteC gateway is not available, call should be routed using HQ gateway.
(3 points)







5.5 Single Number Reach Cisco Mobile Connect

Configure Cisco Mobile Connect feature on HQ Phone 3 2003. Any incoming call to
2003 should ring simultaneously on HQ Phone 3 and HQ PSTN Phone 2722222 and
it can be answered from any of the devices.
Once call is answered from PSTN phone, HQ Phone 3 should display IN Use
Remote mode and call can be successfully switched without losing connection.
Also configure Mobility softkey for HQ Phone 3 which should be used as follows,
1) When there is no active call on HQ Phone 3, mobility feature can be enabled
or disabled using this softkey.
2) When there is an active call on HQ phone 1, mobility softkey can be used to
transfer this call to HQ PSTN phone. When this key is pressed, it should
show Send call to Mobile Phone on IP phone display.

(3 points)

Section 6: Codec Selection

Configure IP Phones and gateways in such as way that all calls within same site
should use G711 codec. Also, all calls between the sites to remote IP phones and
gateways should use G729 codec.
(2 points)









Section 7: Media Resource Management
7.1 IOS Hardware Conference Bridge
Configure DSP resources on HQ router so that HQ IP phones always use
conferencing resources of HQ router first, if available for any type of conference
call made.
Enable 3 conference sessions on HQ router.
(2 points)

7.2 IOS Hardware Transcoding
Configure IOS Hardware transcoding resources in order to meet following
requirements,
1) SiteB IP phones should be able to call ICD Route point number 2400 using
G729 codec.
2) HQ and SiteB IP phones should be able to call Cisco Unity Express voicemail
pilot using G729 codec.
You are allowed to configure maximum three transcoding sessions per router.
Also, you need to configure IOS transcoding only on two routers by looking at the
requirement. If you configure IOS transcoding on all the routers, you will not be
marked for this section.
(3 points)
7.3 MOH
When SiteB and SiteC IP phones or PSTN users are put on hold, configure local
routers to stream G711 multicast MOH from router flash. You can use music-on-
hold.au file in router flash for this multicast requirement.
(3 points)






Section 8: QoS
It is not restricted to use auto-qos however there should not be any impact of the
configuration generated by auto-qos on functionality of the lab. If there is any
such impact, this section will not be marked.
8.1 Switch QoS
1) Map COS 5 to DSCP value of EF
2) On port fa 1/0/13 which is connected to HQ Phone 1, guarantee 32k for
incoming SCCP signaling traffic. Excess traffic should be marked to DSCP 8 and
then transmitted. By default, IP Phones mark SCCP signaling traffic to CS3.
(3 points)

8.2 Link fragmentation and Interleaving
There is 384k frame-relay PVC between HQ and SiteB. Configure R1 and R2 to
enable MLP, link fragmentation and interleaving on this circuit.
(2 points)

Section 9: Voice Mail Integration
You should check MWI functionality for Cisco Unity connection as well as Cisco
Unity Express. Make sure to clear MWI once you test the same in the lab. Also,
make sure that voicemail pilot numbers for both Cisco unity Connection as well as
Cisco unity express are reachable from PSTN.









9.1 Cisco Unity Connection Integration and Configuration
Cisco Unity Connection is pre-configured and integrated with CUCM with following
configuration,
Voicemail Pilot - 2220
Voicemail ports 2221-24
MWI On 1998
MWI off - 1999
AXL username administrator
AXL password ccievoice
Configure users for HQ Phone 1, SiteB phone 1 and SiteB phone 2 in Cisco Unity
Connection. Set default PIN for these users to 246810.
Test the voicemail and MWI functionality for configured users so that call will be
forwarded to voicemail if user does not answer the call within 20 seconds or there
is already an active call on user line.
(2 points)

9.2 Subscriber Customization
HQ Phone 1 user is complaining that he/she is unable to change PIN to 12345
from Unity Connection setup option.
Configure Cisco Unity connection so that HQ Phone 1 user PIN is set to 12345
and either administrator or unity subscriber can modify the PIN.
(3 points)







9.3 Cisco Unity Express Initial Configuration
Cisco Unity Express is set to factory default settings. You need to run through the
initial setup wizard to configure following settings,
IP Address : 142.102.66.253
Hostname : CUE
Domain name : ccievoice.com
DNS : not required
NTP : 142.102.64.254
GUI web administrator : administrator
GUI web password : ccievoice
(2 points)

9.4 Cisco Unity Express configuration and CUCM integration
Configure unity express with following setting and integrate the same with CUCM
cluster.
Voicemail pilot - 4220
Voicemail ports 4221-4223
MWI on 1998
MWI off 1999
Jtapi username cuejtapi
Jtapi password ccievoice
Configure mailboxes for SiteC Phone 1 and Phone 2. Set PIN for these users to
12345. Test the voicemail and MWI functionality.
(3 points)




Section 10: UCCX Applications
UCCX is pre-configured and integrated with CUCM with below details
ICD Route Point 2400
CTI Ports 2401-2405
Jtapi username jtapi
Jtapi password cisco
RmCm username rm
RmCm password cisco
UCCX application username uccxadmin
UCCX application password ccievoice
UCCX server username administrator
UCCX server password - ccievoice

While assigning agents to service queue, it is observed that when ICD pilot point
2400 is called, user is getting message Thank you for callingI am sorry. We are
currently experiencing system problem. Please try again later.
Troubleshoot this ICD application so that when ICD CTI route point is called and
none of the agents are available to handle this call, it should prompt following
message,
Thank you for calling. All our representatives are currently assisting our callers.
Your call is important to us. Kindly stay online and we will assist you shortly.
Do not create new application trigger or any number translation to bypass existing
script. Otherwise, you will not be marked for this section.
(3 points)






Section 11: Cisco Unified Presence

11.1 CUCM presence using busy lamp field (BLF)
Configure 6
th
button of HQ Phone 3 to monitor line status of HQ Phone 1 2001.
When there is an active call on extension 2001, solid red LED should lit on HQ
Phone 3 BLF button. When this BLF button is pressed, call should get connected to
2001.
BLF phone button on HQ Phone 1 6
th
line should display BLF 2001 as label.
(2 points)

11.2 Cisco Unified Presence server and client
Integrate Cisco Unified Presence server with CUCM to achieve following
requirement,
1) Present client installed on client PC 142.100.64.16 should be configured as
a softphone with 2002 as extension number.
2) When there is an incoming call from any IP phone or PSTN number, CUPC as
well as HQ Phone 2 2002 should ring simultaneously. Call can be picked up
from either of the devices.
Do not enable Desktop Phone Configuration on this presence client.
(3 points)










Section 12: High Availability
12.1 SiteB router high availability
Configure SRST on SiteB router so that it provides call processing for all local IP
phones in case of CUCM is not reachable due to WAN issue. Configure following
requirements,
1) Register all IP phones to SRST. Test inbound and outbound PSTN calls. All
IP phones should be able to make 911, long distance and international calls.
Such calls made should display 10-digit caller ID.
2) Enable IP phones to make maximum 2 3-party conference calls.
3) Make sure that voicemail functionality is restored in event of WAN failure.
Voicemail forwarding feature should work between IP phones as well as
PSTN calls. When such forwarded call comes to Cisco Unity connection, it
should play users personal greeting. You are not allowed to use alternate
extension to achieve this.
(3 points)

12.2 CUCM Call forward unregisterd
Make sure that all HQ IP phones should be able to call 3001 using 4-digit dialing in
event of WAN failure.
(2 points)

12.3 SiteC High Availability

Configure SRST on SiteC router so that it provides call processing for all local IP
phones in case of CUCM is not reachable due to WAN issue. Configure following
requirements,
1) Register all IP phones to SRST. Test inbound and outbound PSTN calls. All
IP phones should be able to make 999 emergency, and local calls. Such calls
made should display 8-digit caller ID 2404xxxx.


2) Make sure that voicemail functionality is restored in event of WAN failure.
Voicemail forwarding feature using local Cisco unity express should work
between IP phones as well as PSTN calls. For forwarded calls, it should play
users personal greeting. Test the voicemail from PSTN and MWI
functionality in event of WAN failure.
(4 points)