CCIE Collaboration v1 - Question Set - Final Release - 10-08-2016 - Lab 3

WWW.CCIECOLLABLABS.
COM FINAL RELEASE LAB 3:25-MAY-16
QUESTIONS SET
LAB 3
www.cciecollaborationlabs.com
WWW.CCIEVOICELABS.COM WWW.CCIECOLLABORATIONLABS.COM
WWW.CCIECOLLABLABS.COM FINAL RELEASE LAB 3:10-AUG-16
Diagram 1:
IP Network Topology
Diagram 2:
Telephony Topology and Numbering Scheme
PSTN PHONE:
Diagram 3:
PORT/INTERFACE CONNECTIONS
HQ-SW 1
Gigabit Ethernet 1/0/1 R1 Gi0/0
Gigabit Ethernet 1/0/2 HQ CUCM PUB, HQ CUC
Gigabit Ethernet 1/0/3 HQ IM&P, HQ UCCX
Gigabit Ethernet 1/0/4 HQ CUCM SUB, PC-1
Gigabit Ethernet 1/0/13 HQ Phone 1 - 7965
Gigabit Ethernet 1/0/14 HQ Phone 2 - 9971
Gigabit Ethernet 1/0/24 BB – L2sw2 = Trunk allowed 100 – 102 – 202
DO NOT TOUCH
HQ-R1
Gigabit Ethernet 0/0 SW1 Gigabit Ethernet 1/0/1
Gigabit Ethernet 0/1 (157.26.1.254) BB L2sw2 Gigabit Ethernet 1/0/7
Serial 0/1/0.101 R2 Serial 0/2/0.101
Serial 0/1/0.201 R3 Serial 0/2/0.201
Site B - R2
Serial 0/2/0.101 R1 Seria0/1/0.101
Gigabit Ethernet 0/0/0 SiteB Phone 1 - 7965
Gigabit Ethernet 0/0/1
Gigabit Ethernet 0/0/2 Site B CUCM Pub, Site B IM&P& PC-2
Gigabit Ethernet 0/0/3 BB – L2sw2 = Trunk allowed 1,302,402
DO NOT TOUCH
Site C - R3
Serial 0/2/0.201 R1 Seria0/1/0.201
Gigabit Ethernet 0/0/0 SiteC Phone 1 - 7965
Gigabit Ethernet 0/0/1 SiteC Phone 2 - 9971
Gigabit Ethernet 0/0/2 BB – L2sw2 = Trunk allowed 1,502,602
DO NOT TOUCH ccie voice labs dot com
Service Module 1/0 Cisco Unity Express
Diagram 4:
Router Interface IP address
SW1
VLAN 202 142.202.64.253
R1
Gigabit Ethernet 0/0.100 142.100.64.254
Serial 0/1/0.101 156.26.101.1
Serial 0/1/0.201 156.26.201.1
Gigabit Ethernet 0/1 156.26.1.254
Loopback0 142.1.64.254
R2
VLAN 200 SERVER VLAN 142.100.65.254
VLAN 302 VOICE VLAN 142.102.65.254
VLAN 402 DATA VLAN 142.202.65.254
Serial 0/2/0.101 156.26.101.2
Loopback0 142.1.65.254
R3
VLAN 502 VOICE VLAN 142.102.66.254
VLAN 602 DATA VLAN 142.202.66.254
Serial 0/2/0.201 156.26.201.2
Loopback0 142.1.66.254
Diagram 5:
SERVER, PC & MISC IP ADDRESS
HQ SB
HQ - PUB 142.100.64.11 SB PUB 142.100.65.11
HQ – SUB 142.100.64.12 -
HQ – CUC 142.100.64.13 SB CUC 142.100.65.13
HQ – UCCX 142.100.64.14 -
HQ – IM&P 142.100.64.15 IM&P 142.100.65.15
PC-1 142.100.64.21 PC-2 142.100.65.21
BACKBONE
BB – TMS (W2008 Server) 157.26.1.8
BB – CUCM Publisher 157.26.1.11
BB – VCS 157.26.1.15
NTP 157.26.1.250
Alternate SIP Server 157.26.1.253
Candidate PC 152.YY.100.10
Candidate Term-Server 152.YY.100.1
Rack Number YYcciecollaborationlabs d o t com
Diagram 6:
User Details
Please use the floowing user information for all users in CUCM, CUCME, Cisco Unity Connection, CIsco
Unity Express, wherever applicable.
User id First Name Last Name PIN PASSWORD

hqone one hq 12345 cisco
hqtwo two hq 12345 cisco
sitebone one siteb 12345 cisco
sitecone one sitec 12345 cisco
sitectwo two sitec 12345 cisco
uccxadmin uccx admin 12345 cciecollab
grade DO NOT Touch DO NOT CHANGE
General Guidenlines:
A new Phone Load is now specifically mentioned for HQP2. This load IS NOT active on the HQ Phone2
at the start of the lab. Proctor will instruct you to change firmware load to the new one. This NEW
FIRMWARE is inactive load in the CUCM. Changing this will solve the Host Not Found issue for One
Button Login.KINDLY NOTE THIS IS A PART OF CCIECOLLABLABS . COM
If you do show cdp nei, you will not be able to see the mac address of 9971 phones. You can get the
mac address of the 9971 Phones using the show mac-address-table.
PSTN Dialing Instructions:

The PSTN Phone is used place/receive calls to/from each of your sites. When calling from PSTN
Phone to any of your site DID number, please remember the following rules.
1. You do not need to dial any leading Digit.
2. Dialing from PSTN Phone to Site DID number is localized. Please see point 3,4& 5 below.
3. Line 1 and 2 assumes US dialing patterns

1. 7 digits for local calls
2. 1 + <10 digits> for long distance calls only
3. 011 + <country code> + <any number of digits> for international calls
4. Line 4 and 6 assumes Hong Kong and U.K dialing patterns

1. 8 digits for local calls
2. 00 + <country code> + <any number of digits> for international calls
5. For example, to call, HQ Phone 1 (+14082022001) from PSTN, you need to dial
1. 2022001 if you use line 1, since line 1 belongs to local PSTN at HQ
2. 14082022001 if you use line 2, since line 2 is placing a US long distance call to
reach the HQ area code
3. 0014082022001if you use line 4 or line 6, since these 2 lines is placing a
international call to U.S.
6. Similarly, to call, SC Phone 1 from PSTN, you need to dial

1. 01185224044001 if you use line 1, since line 1 or 2 (U.S. PSTN)
2. 24044001 if you were dialing from line 4 (Local Hong Kong PSTN)
3. 0085224044001 if you were dialing from line 6 (U.K. PSTN)
Section 1:
CONFIGURE & TROUBLESHOOT CISCO COLLABORATION INFRA
1.1 Voice and Data VLANs
Configure Voice VLANs for your IP Phones at the HQ, SiteB and Site C. The voice VLAN numbers are
102 for HQ, 302 for Site B and 502 for Site C. Please refer to Topology Diagram and the Port
Assignment tables for detailed information.
There will be also PCs connected to the PC port of these IP phones. Configure the switch ports to
place these PCs into the data VLAN, which are 202 for HQ, 402 for Site B and 602 for Site C.
Score:
2 points
1.2 DHCP & NTP Services
Configure the HQ Cisco Unified Communications Manager Subscriber server at 142.100.64.12 as the
DHCP server to assign IP addresses to IP phones at the HQ Site. HQ IP phones need IP addresses from
the local voice subnet (142.102.64.X/24) ranging from 142.102.64.30 to 142.102.64.50.
Configure the Site B Cisco Unified Communications Manager Publisher server at 142.100.65.11 as the
DHCP server to assign IP addresses to IP phones at the Site B. Assign addresses from the local voice
subnet (142.102.65.X/24) in the range of 142.102.65.30 to 142.102.65.50 to Site B phones.
Configure the Site C 2921 router (R3) as a Cisco IOS DHCP server, which assigns IP addresses to the IP
phones at the local voice VLAN. IP phones at this site need IP addresses from the local voice subnet
(142.102.66.X/24) ranging from 142.102.66.30 to 142.102.66.50.
Also R1,R2 and R3 to synchronize local clock with a backbone NTP server at 157.26.1.250. The
Backbone NTP server is running on coordinated Universal Time. Configure HQ CUCM Publisher to
synchronize clock with R1 at 142.1.64.254, while all other applications should synchronize their clock
to the backbone NTP server.
HQ CUCM and R1 are in the North American Pacific Standard time zone, which is eight hours behind
UTC time. Site B CUCM and R2 are in the North Central Standard Time zone, two hours ahead of HQ
time. Also configure Site C router (R3) in Hong Kong to synch time with the backbone NTP server
which is eight hours ahead of UTC time. Once registered, the IP Phones at each site should display
the correct local time
Score:
3 points
Section 2:
Configure & Troubleshoot Cisco Unified Communications Manager
2.1 CUCM IP Phones
2.1a-CUCM: SCCP IP PHONE
Register HQ phone 1 to HQ CUCM cluster and Site B phone 1 to Site B CUCM, according to provided
telephony number scheme. Ensure the HQ phone is provisioned with TFTP redundancy such that the
publisher CUCM is the backup TFTP server. Extension-to- extension dialing of each site uses 4 digits
only. Caller name display should be delivered on internal 4- digit calls- use trivial caller names such
as “HQ Phone 1” and “Site B Phone 1”. Lastly, both phones should display globalized calling number
on the upper right hand corner of the phone screen. Refer to the exhibit below for an example
captured on HQ Phone 1.
Score :
2 Points
2.1b-CUCM: SIP IP PHONE
Register and ConfigureHQ Phone 2 as SIP phone to the HQ Cluster. Ensure the phone is provisioned
with TFTP redundancy such that the publisher CUCM is the backup TFTP server. Caller name display
should be delivered on internal 4-digit calls- use trivial caller names such as “HQ phone 2”. Exclusive
from cciecollaborationlabs.com. Enable the video camera on HQ Phone 2. Use G.722 codec for
internal calls with the other HQ Phone. Globalized calling number should be displayed on the top of
the phone screen. Refer to the exhibit below for an example captured as HQ Phone 2.
Score:
2 Points
2.2 CUCM Gateways
Information below is relevant to your digital T1 gateways.
 Line coding/framing b8zfs/esf

 ISDN switch type Primary- NI
 Configure your gateways to take clock (layer1) from network. For PRI circuits configure your
gateway as Layer 2 User side.
 Refer to the individual gateway sub-sections for digit sending and receiving details.
 Calling names should always be sent to the PSTN
 Use full span on T1 PRI unless specified otherwise.
 Gateway configuration must be verified with successful completion of inbound and outbound
calls. Therefore, simple route pattern configuration is expected, even though call routing is
not the key focus of this section.
2.2a HQ CUCM: MGCP Gateway
Configure and register R1 as IOS T1 PRI MGCP gateway to the HQ CUCM cluster. All MGCP traffic
should use R1 loopback interface. Implement CUCM redundancy such that if the primary call agent
(CUCM subscriber) goes down, the back up call agent (CUCM publisher) should take over.
Telco delivers 10 digit direct inward dialing (DID) called numbers in inbound calls, 408202xxxx, where
xxxx is any HQ internal 4-digits extension.
Verify you HQ MGCP gateway configuration by placing an inbound call from line 1 on the PSTN
phone to 202xxxx, where xxxx is any HQ internal 4-digit extension.
For outbound calls, ensure 911 calls from any HQ IP phone terminate on R1. For 911 calls, send
408202xxxx, where xxxx is the 4-digit extension as the calling number. 9911 is not required. Please
also refer to the dial plan section for all call routing requirements.
Calls traversing between HQ phone and R1 PRI trunk should always use G711mulaw codec. Also
ensure the following verification command is produced on R1
R1# show ccm-mamanger host
Score:
2 Points
2.2b-Site B CUCM: H.323 Gateway
Configure and register R2 as IOS T1 PRI H.323 gateway for the Site B CUCM Cluster, All H.323 traffic
should use R2’s loopback interface.
Telco delivers 10-digits direct inward dialing (DID) called numbers on inbound calls 972303xxxx,
where xxxx is any SiteB internal 4-digits extension.
Verify your SiteB gateway configuration by placing an inbound call from PSTN phone line 2 to
303xxxx, where xxxx is any SiteB internal 4-digit extension.
For outbound calls, ensure any SiteB IP phone call 911 through R2. For 911 calls, send 972303xxxx,
where xxxx is the 4-digit extension, as the calling number. You are not required to configure 9911.
Please also refer to the dial-plan section for all call routing requirements.
Calls between Site B IP phone and R2 PRI trunk should always use G722-64 codec.
Score:
2 Points
2.3 CUCM SIP Trunks
2.3a-Inter-Site SIP Trunk using CUBE: HQ & Site B
Configure R2 as Cisco Unified Border Element interconnecting HQ and SiteB CUCM clusters using SIP
protocol, which must terminate media traffic but negotiate codecs with the endpoints. Make sure
iLBC codec is use for all 4-digit inter-site calls between HQ and Site B (except for Unity Connection
calls,refer to section 5 for details). For example , when 2001 calls 3001, the call must go through R2
using iLBC codec.
Score:
3 Points
2.3b-CUCM SIP Trunk to PSTN Troubleshooting
Build a SIP trunk between the HQ CUCM cluster and the backbone CUCM (IP address 157.26.1.11).
The backbone CUCM is expecting the SIP calls delayed offer.
Create a single route pattern 85151111,on your CUCM and send the call over the SIP Trunk to the
backbone CUCM. Use 8202xxxx, where xxxx is the 4-digit internal extension number, as the calling
number for these calls. Do not send the calls through any intermediate routing entity (such as CUBE).
Please test calls to the backbone CUCM from HQ Phone 2. These calls will fail. Use RTMT and CUCM
traces to troubleshoot the call failures.Enable the HQ CUCM to send periodic (every 30 seconds) SIP
packets to the Backbone CUCM to ensure IP reachability.
Find the following two SIP events and save the exact SIP messages in two separate notepad files on
PC-1’s Desktop, and name the files ‘SIP-TS-EVENT-1.txt’ and ‘SIP-TS-EVENT-2.txt’ correspondently.
 Event 1: The SIP message HQ Phone 2 received which triggered the busy tones.
 Event 2: The SIP message from the backbone CUCM in response to the periodic reachability
checks.
Score:
4 Points
2.4 CUCM DIAL PLAN
General Dial-plan Rules:
Please note that due to the timed nature of the lab exam, this section is not designed to build a
comprehensive and fully redundant dial plan between all sites.
Instead, Candidates are expected to demonstrate their knowledge and experience by fulfilling a set
of specific call-routing requirements.
Call-routing configurations beyond these specific requirements will not be marked and will not result
in additional points.
Do not us the “@” wildcard in your CUCM dial plan.
Country codes used in the lab exam are “1” (United States), “44” (United Kingdom), and “852” (Hong
Kong).
The PSTN access code at all sites is “9”. Country’s international access code is “011” for U.S and “00”
for Hong Kong.
Read the entire Call Routing section and understand all dialing requirement, before proceeding to
configure your dial plans.
2.4a-HQ PSTN Dial plan
The following call routing policies are in effect at HQ's PSTN Service Provider
1) HQ PSTN service provider mandates proper of both ISDN “called party number” and
“Called party number type” (Subscriber, National and International)
2) Calling and Called party number types(Subscriber, National, International) must be set in ISDN
messages for different types of calls (Local, National and international)*/
3) Do not send leading digits in the ISDN called party number string, such as "1" for national and
"011" for international to signal type of calls.
Configure HQ CUCM to satisfy the following dial plan requirements. Access code for all PSTN call is 9
 Configure local route group for HQ's local Voice gateway : R1
 All HQ IP phones should be able to call local 7 digits PSTN numbers by dialing the access code
"9" followed by 7 additional digits. The First digit after the access code could be any digit
from 2 to 9, the remaining digit could be 0 (zero) to 9. This type of call should always use the
local gateway R1 as the primary route. 7-digit calling number (202xxxx where xxxx is internal
extension) and calling name should be sent to PSTN.
 All HQ IP Phones should also be able to place a national long Distance PSTN call by dialing
access code “9”, followed by “1”, followed by a 3-digit area code, and lastly a 7-digits
subscriber number. The first digit of subscriber number could be any digit from 2 to 9, the
remaining area code digits and the subscriber number digits could be 0 (zero) to 9. This type
of call should always use the local gateway R1. 10 digit calling number (408202XXXX where
xxxx is internal extension) and calling name should be sent to the PSTN. Use 91206765xxxx
where xxxx can be any digit combination, to test your HQ long distance calls, the calls will ring
PSTN line 1.
 When HQ users place a call to Site B PSTN number, i.e.9725252222, the call should be routed
via SIP to the CUBE at R2, and subsequently be forwarded to Site B CUCM to terminate as a
local call for Site B. All rules for question 2.3a apply for this type of calls.
 All HQ IP phones should also be able to place international calls by dialing the access code "9"
followed by "011", followed by variable length digits, with or without ‘#’ sign at the end.
This type of call should always use local gateway R1. +1408202xxxx where xxxx is internal
extension should be sent as calling number. Calling name should be sent to PSTN.
All the above mention calls in the question must be serviced by single route pattern”\+!”. The
exception to this rule is SIP Trunk calls.
Score:
4 points
2.4b-Site B PSTN Dial plan
The following call routing policies are in effect at Site B's PSTN Service Provider
1) SiteB PSTN service provider mandates proper formation of both ISDN “called party number” and
“Called party number type” (Subscriber, National and International)
2)Calling and Called party number types(Subscriber, National, International) must be set in ISDN
messages for different types of calls (Local, National and international)
3) Do not send leading digits in the ISDN called party number string, such as "1" for national and
"011" for international to signal type of calls.
Configure SiteB CUCM to satisfy the following dialplan requirement. Access code for all PSTN call is 9
 Configure local route group for Site B's local Voice gateway : R2
 All Site B IP phones should be able to call local 7 digits PSTN numbers by dialing the access
code "9" followed by 7 additional digits. The First digit after the access code could be any
digit from 2 to 9, the remaining digit could be 0 (zero) to 9. This type of call should always use
the local gateway R2 as the primary route. 7-digit calling number (303xxxx where xxxx is
internal extension) and calling name should be sent to PSTN.
 All Site B Phones should also be able to place a national long distance PSTN call by dialing
access code “9”, followed by “1” followed by a 3-digit area code, and lastly a 7-digit
subscriber number. The first digit of Subscriber number could be any digit from 2 to 9, the
remaining area code digits and the subscriber number digits could be 0 (zero) to 9. This type
of call should always use the local gateway(R2). 10 digit calling number (972303XXXX where
xxxx is internal extension) and calling name should be sent to the PSTN.
 When Site B users place a call to HQ PSTN number, i.e. 4085151111, the call should be routed
via SIP to the CUBE at R2, and subsequently be forwarded to HQ CUCMs to terminate as a
local call for HQ. All rules for question 2.3a apply for this type of calls.
 All Site B IP phones should also be able to place international calls by dialing the access code
"9" followed by "011", followed by variable length digits, with or without # sign at the end.
This type of call should always use local gateway R2. +1972303xxxx where xxxx is internal
extension should be sent as calling number. Calling number should be sent to PSTN.
All the above mention calls in the question must be serviced by single route pattern”\+!”. The
exception to this rule is SIP trunk calls.
Score:
3 points
2.4c-SiteB Mobile Voice Access

Enable Mobile Voice Access to allow the owner of Site B Phone 1 to originate a call from his/her
mobile phone, 9725252222, as if it was dialing from the desktop phone.
Provision a DID number, 9723033300, for Site B Phone 1 owner to access the enterprise voice
network from the mobile phone. When a call is placed from the mobile number to 9723033300 or
3033300, Site B Phone 1 owner will be asked to authenticate using the user pin number (12345).
Once authenticated, the owner should be able to place Calls to any destination allow on his/her
desktop phone, except the emergency 911 number.
Ensure the correct calling numbers are sent to all destinations.
Score:
4 points
2.4d-Site B Plus Dialing
Configure Site B CUCM to deliver the following globalized calling number to SiteB IP phones.
1) When HQ PSTN (PSTN Phone Line 1) calls Site B IP Phone by dialing 19723033001 and the call
is not answered, the "missed calls" directory on Site B Phone 1 should display the missed call
in a globalized format with “+” sign. " +14085151111".This call should be routed via SIP to
CUBE at R2,and subsequently be forwarded to HQ CUCMs to terminate as a local call for HQ.
All rules for question 2.3a apply for this type of calls.
2) When Site C PSTN calls Site B IP Phone by dialing 0019723033001 and the call is not
answered, the "missed calls" directory on Site B Phone 1 should display the missed call in a
globalized format with “+” sign. " +85225353333".
3) For any of the missed calls mentioned above, the owner of the Site B phone 1 should be able
to press the "dial" soft key to place a call to appropriate destination through the Site B
Gateway (R2), along with calling name.
Score:
3 Points
2.4e-HQ SIP Trunk redundancy to PSTN
In Question 2.3b you build the SIP Trunk between the HQ CUCM cluster and the backbone CUCM (IP
address 157.26.1.11).
Since calls through this SIP trunk failed, the VoIP Service provider has enabled an alternate SIP trunk
at 157.26.1.253, for you to try the same call. The route pattern is the same 85151111, however you
must now originate your SIP request from R1 with IP address 142.102.64.254.
Do not delete the first SIP Trunk from your HQ CUCM. Calls to 85151111 should always go to
157.26.1.11 first. Route the call to alternate SIP trunk only when calls could not complete through
157.26.1.11. Continue to use 8202xxxx where xxxx is the 4-digit internal extension number and the
calling number for these calls.The call through R1 should complete and bi-directional video should be
established
Enable periodic (every 30 seconds) SIP packets to the alternate SIP trunk to ensure IP reachability
Score:
3 Points
2.5 CUCM: URI Dialing
2.5a-HQ Intra-site URI Dialing
Create a URI for HQ Phone 1 (hqone@cciecollab.cisco.com) and HQ Phone 2

(hqtwo@cciecollab.cisco.com) in the Directory URI partition. Make the 5th button on each phone, an
URI speed dial button to the other phone and ensure the speed dial works between the two
phones.The Speed dial button on each phone should display the full Directory URI string. When HQ
Phone 2 receives a call from HQ Phone , it should display URI as well as the caller name.
Score:
3 Points
2.5b-HQ & SiteB Inter-site URI Dialing
Configure ILS between HQ & Site B CUCM clusters. Set HQ CUCM as an ILS network hub while the
Site B CUCM is a spoke. Make sure the clusters check for updates every minute.
The Directory URI for Site B Phone 1 is sitebone@cciecollab.cisco.com Add two speed dial entries on
the 5th and 6th button of Site B Phone 1, the 5th button should call hqone@cciecollab.cisco.com and
the 6th button should call hqtwo@cciecollab.cisco.com
Add an additional 6th button on each HQ phone for Site B Phone 1’s URI.The speed dial button on
each phone should display the full Directory URI string. When Site B phone 1 speed dials HQ Phone 2,
its URI and caller name should be delivered on HQ Phone 2. URI calls between HQ and Site B should
use iLBC codec.
Score:
3 Points
2.6 CUCM: Media Resources
2.6a-HQ IOS Video Conference Bridge: R1
Configure and register R1 as Cisco IOS Video Conference Bridge for the HQ CUCM cluster.
HQ Phone 2 should be able to invoke this conference bridge for a three way video conference using
H.264 codec.
After you properly register and provision routing for Site C Phone 2, make sure to verify that HQ
Phone 2 can host a Video Conference with PSTN phone and Site C Phone 2. From HQ Phone 2, first
place a call to PSTN Phone line 1 (85151111) and then conference in Site C Phone 2 at 4002.
Score:
4 Points
Section 3:
Configure & Troubleshoot CISCO IOS UC Application & Features
3.1 CUCME IP PHONES
3.1a-CUCME SCCP IP PHONES
Register and Configure Site C Phone 1 according to the telephony number scheme. This phone should
display globalized calling number, “+85224044001”, on the top right hand corner of the phone
screen. Extension-to-Extension dialing at each site uses the last 4 digits only. Caller name Display
should be delievered on the internal 4-digit calls: use trivial caller names such as “Site C Phone 1”.
Make sure the phone displays the correct local time and that the date display is arranged as
dd/mm/yy.
Score:
2 Points
3.1b-CUCME SIP IP Phones
Register and configure Site C Phone 2 according to the provided telephony number scheme.
This phone should display globalized calling number, “+85224044002”, on the top of the phone
screen. Extension-to-Extension dialing at each site uses the last 4 digits only. Caller name display
should be delivered on internal 4-digits calls: use trivial caller names such as “Site C Phone 2”. Make
sure the phone displays the correct local time and that the date display is arranged as dd/mm/yy.
Lastly enable the video globally on SIP phones.
Score:
2 Points
3.1c-CUCME IP Phone Shared Lines
Assign 4001 to the second line on the Site C Phone 2. Both lines should ring whenever the directory
number is called within Site C, across from the SIP trunk, or from the PSTN.
Score:
3 Points
3.2 CUCME Gateway
Information below is relevant for your digital E1 gateway.
 Line code/framing HDB3/CRC4

 ISDN switch type Primary- 5ess
 Configure your gateways to take clock (layer1) from network. For PRI circuits, configure your
gateway as Layer 2 User side.
 Refer to the individual gateway trunk-section for digit sending and receiving details.
 Calling names should always be sent to the PSTN
 Use full span on E1 PRI unless specified otherwise.
 Gateway configuration must be verified with successful completion of inbound and outbound
calls. Therefore, simple route pattern configuration is expected, even though call routing is
not the key focus of this section.
3.2a-Site C E1 – PRI Trunk
Configure E1 PRI on the Site C 2921 router R3, provision this router as H.323 gateway. Use only the
first 12 channels on this PRI circuit. This gateway should send and receive all H.323 traffic from its
local voice VLAN interface with IP address of 142.102.66.254. Telco delivers 8-digit Direct-Inward-
Dial (DID) on inbound calls.
Verify your configuration by placing an inbound call from PSTN phone line 4 to any DID numbers
(2404xxxx) at Site C, where xxxx is any Site C internal 4-digit extension.
Also make sure the outbound calls work by calling the local emergency number (999) for any Site C IP
phones. For emergency calls, send 2404xxxx, where xxxx is 4-digit extension, as the calling number.
Score:
2 Points
3.3 CUCME PSTN Dial plan
3.3a-Site C PSTN Dial Plan
Site C PSTN Service Provider routing policies.
 Site C PSTN Service Provider requires proper setting of called party number string and called
party number types (subscriber or international) for different types of calls (local or
international)
 Called Party number types (subscriber or international) must be set in ISDN setup messages.
 For example, if Site C IP phones dials an U.S. international number, 90014085151111, Site C’s
PSTN would process the call only if “14085151111” was sent as the called number
accompanied by called party number type of “International”.
 “Unknown” called party number types is only accepted for emergency 999 calls.
After you understood the Site C PSTN routing rules above, configure the following dial plan
requirements for Site C.
1> All Site C IP phones should be able to place local PSTN calls by dialing the access code 9
followed by any 8 digit number string. These local calls should go out of local H.323 gateway
R3. Send calling name display and 8-digit calling party number (2404xxxx) to PSTN. Calling
party number type for these calls should be set to subscriber.
2> All Site C IP phones should be able to place international calls by dialing the access code 9,
followed by 00, and then followed by country code and variable length digit patterns. For
example, Site C users should dials 900442085554444 to reach the UK PSTN number. Some
users dial ‘#’ to signal end-of-dial and to avoid having to wait for the interdigit time out, while
others do not. Send all international calls to R3. For international calls, send calling party
number with Site C’s country code with (+8522404xxxx) and calling party number type as
“International”, calling name should also be sent.
Score:
2 points
3.3b-Site C Inter-site Dial Plan to HQ & Site B
Configure R2 as Cisco Unified Border Element interconnecting HQ and Site B CUCM clusters with Site
C using SIP protocol. All 4-digit inter-site calls between Site C and HQ or Site B must go through R2
using SIP. For example, when Site C phone 2 calls Site B Phone 1 by dialing 3001 or vice versa, the
calls signal and media should traverse R2, which terminates media traffic but pass-through codecs
requested by the endpoints.
Intersite voice calls between Site C and other two Sites should use G.729 codec. Use a Single voip
dial-peer on R3 to route the calls to HQ and Site B. Lastly make sure h.264 video works between Site
C Phone 2 and HQ Phone 2.
Score:
3 Points
3.4 CUCME IP Phone Services and Media Resource
3.4a-Site C Hardware Ad-Hoc Conference Resources
Register a Hardware Conference Bridge to the R3 to service Site C local Phones.Either phones should
be able to invoke ad-hoc confercences with the PSTN as well as 4-digit numbers at the other two
sites. Provision a conference button with functions like the conference soft key, on the sixth line of
Site C Phone 1. Also make sure the Site C Phone can select active or on hold calls and add them to a
conference.
Score:
3 Points
3.4b-Site C IP Phone Customization
Site C IP Phone users have been opening customer service request on how to use the extension
mobility on the IP Phone service menu. Since extension mobility is not yet configured, the
management has decided to remove this option from the IP Phone service menu. You have been
asked to implement this change on all Site C IP Phones.
Score:
3 Points
3.5 Cisco Unity Express
3.5a-Site C CUE Initialization
Your Cisco Unity Express Service engine has been restored to factory default. You will need to go
through the one time service engine post installation configuration tool to initialize.
Use the following information for Cisco Unity Express Initialization and make sure CUE is reachable
through your network.
IP Address: 142.1.66.253
Hostname: CUE
Domain name: cciecollab.cisco.com
DNS: not necessary
NTP: 157.26.1.250
Time Zone: Hong Kong
CUE web GUI account: administrator
CUE web GUI account password: cciecollab
Score:
2 Points
3.5b-CUE and CUCME Integration
Integrate CUE and CUCME using following information.
 Voice mail pilot number: 4220

 Set all voice mail account pins to: 12345
Create account for Site C Phone 1 and Phone 2. Make sure messages can be deposited and MWI
works for local IP phones. Call should be forwarded to voicemail if the extension is busy or if an
incoming call was not answered in 20 seconds. Make inbound PSTN and intersite 4 digits calls can be
forwarded to CUE with message deposit and MWI functions. When Site C Phone 1 user calls into CUE
from his/her cell phone, +85225353333, CUE should directly prompt pin number as if the call is
coming from Site C Phone 1.
Score
3 Points
Section 4:
Configure & Troubleshoot QOS & Security in Cisco Collaborations
Solutions
4.1 QOS: Classification and Mapping
4.1a-Switch COS to DSCP Mapping
Configure SW1 and Ethernet Switch module on R2 and R3 for the following COS to DSCP mapping for
voice and interactive video traffic.
 Voice signaling CoS 3 to CS3

 Voice Media CoS 5 to EF
 Video Media CoS 4 to AF41
Score:
3 Points
4.2 QOS: Congestion Management
4.2a-Switch Traffic Marking and Policing
Enable the following Voice and Video traffic management policies on voice ports with IP phone
attached.
 Allow a maximum of three G.711 calls. Any excess voice media traffic should be dropped
 Allow 24 kbps for voice signaling (SCCP and SIP) . Any excess signal traffic should be marked
down to CS1.
 Allow 5 MBPS for Video media, excess video media traffic should be marked down to CS1.
 All data traffic should be remarked to best effort
Score:
4 Points
SECTION 5:
Configure and Troubleshoot Cisco Unity Connection
5.1 Cisco Unity Connection: Integration
5.1a-CUC SCCP Integration
Integrate the HQ CUCM cluster with HQ CUC using SCCP, with the following information
 Voicemail Pilot – 2220

 Voicemail ports – 2221-2223
 MWI On DN – 1998
 MWI Off DN – 1999
Set the voicemail pilot number to 2220. After the Integration is completed, you should be able call
into the voice mail pilot number from any HQ Phone. Make sure the local PSTN Phone can also place
a call from Line 1 to 2022220 and hear the CUC system greeting. All Internal calls to CUC should use
G.711 codec.
Score:
2 Points
5.1b-CUCM SIP Integration
Integrate the Site B CUCM and HQ CUC using SIP. Set the Voicemail pilot number to 3220. After the
integration is completed you should be able to call into the voicemail pilot number from any Site B
Phone extensions. Make sure the PSTN phone can also place a call to 9723033220 and hear the CUC
system greeting. All calls from Site B to CUC should use G729 codec.
Score:
3 Points
5.2 Cisco Unity Connection: Administration
5.2a-Voicemail Provision for HQ users
Create users for HQ Phone 1 (hqone) and HQ Phone 2 (hqtwo) on CUC. Set passwords to be “12345”.
Also make sure Incoming calls to these phones, when not answered in 20 seconds or when there is
an active call on the line, are forwarded to Cisco Unity Connection for Voicemail Service.
Verify that you can leave and retrieve messages, and that MWI light works.
Score:
3 Points
5.2b-Voicemail Provision for Site B users
Import the “sitebone”user from CUCM into CUC. Incoming calls for Site B Phone 1, when not
answered in 20 secs or when it is engaged in an active call, should be forwarded to CUC. For security
purposes, the site B user prefers to have CUC announcing his/her extension number (3001) instead
of his/her name or alias, do not use recorded name to achieve this. Verify that you can leave and
retrieve messages and that MWI light works.
Score:
3 points
Section 6:
Configure and Troubleshoot Cisco Unified Contact Centre Express
6.1 Cisco Unified Contact Centre Express: Integration
6.1a-HQ UCCX Integration
Configure the following CUCM, UCCX and ICD script to accommodate the customer requirements for
HQ listed below
CTI Route Point – 2400

CTI Ports – 2401 to 2405
Jtapi Prefix - jtapi
Jtapi password – cisco
Rm username – rm
Rm password – cisco
Add a second line on HQ Phone 1 (DN 2201) to be dedicated to agent calls. Configure the IP phone
agent phone service for this phone. Configure the ICD so that calls entering into the queue can hear
how many calls are in currently in the queue. For example, if there is one call waiting in the queue,
another new call into the queue will hear “Thank You for your patience, there are currently two calls
waiting, our customer service respresentative will answer all calls in the order they were received.”
Score:
4 Points
Section 7:
Configure & Troubleshoot Cisco Unified IM & Presence
7.1 IM&P: Integration
7.1a-HQ IM&P integration: CSF Soft Phone with Video
Integrate HQ CUCM with IM and Presence services. Provision the jabber client on PC1 for HQ user
“hqone”. Make sure the client can place audio phone calls with video using the high resolution video
camera attached to the PC-1. Place a call to HQ Phone 2 and you should see the video streams
between two endpoints.
RDP into PC1 with the following credentials username” admin” password ”cciecollab”
Refer to the following screen shot for provisional and sign on information. The jabber client should
be able to search the CUCM Directory to find contacts.
Score:
4 Points
7.1b-HQ IM&P Integration: Desktop Phone Control
Provision the jabber client on PC 2 for HQ user“hqtwo”. The user should be able to use the Client to
control the Desk 9971 HQ Phone 2 or, as a SoftPhone. When in Softphone mode, user should be able
to share desktop with another client.
RDP into PC2 using the following credentials user name “admin” password“cciecollablabs”
Refer to the following screen shot for provisional and sign on information
Score:
3 Points
ALL OUR ACTIVE CLIENTS CAN REQUEST FOR FINAL CONFIGURATIONS / BASE CONFIGURATIONS/
IPCC SCRIPTS FROM US AT SUPPORT@CCIECOLLABORATIONLABS.COM
OUR CCIE COLLABORATION ENGINEERS ARE AVAILABLE ON GOOGE TALK CHAT for support any
questions related to our workbooks at (cciecollaborationlabs@gmail.com)
http://cciecollaborationlabs.com/contactus.asp (LIVE SUPPORT)
http://cciecollaborationlabs.com/cciecollaborationupdates.asp (UPDATED DATE)
YOUR GATEWAY TO SUCCESS TOWARDS CCIE LAB
ACTIVE CLIENTS WILL GET VERY SPECIAL DISCOUNTS ON OTHER CCIE TRACKS
KINDLY VISIT FOR FURTHER INFORMATION
CCIE SECURITY ---->WWW.CCIESECURITYLABS.COM (CSL)
CCIE WIRELESS ---->WWW.CCIEWIRELESSLABS.COM (CWL)
CCIE DATACENTER ---->WWW.CCIEDATACENTERLABS.COM (CDL)
CCIE COLLABORATION ---->WWW.CCIECOLLABORATIONLABS.COM (CCL)
CCIESERVICEPROVIDER.COM ----->WWW.CCIERNSLABS.COM (CRL)
CCIESERVICEPROVIDER.COM ----->WWW.CCIESERVICEPROVIDERLABS.COM (CSPL)
WORLD FIRST REAL RACK RENTAL FOR ALL THE CCIE TRACKS ----> HTTP://CCIERACK.RENTALS
KINDLY CONTACT US AT SALES@CCIECOLLABORATIONLABS.COM FOR FURTHER INFORMATION ON

OTHER TRACKS
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WWW.CCIECOLLABLABS.

COM FINAL RELEASE LAB 3:25-MAY-16

Telephony Topology and Numbering Scheme

Router Interface IP address

SERVER, PC & MISC IP ADDRESS

User id First Name Last Name PIN PASSWORD

PSTN Dialing Instructions:

1. You do not need to dial any leading Digit.

3. Line 1 and 2 assumes US dialing patterns

4. Line 4 and 6 assumes Hong Kong and U.K dialing patterns

6. Similarly, to call, SC Phone 1 from PSTN, you need to dial

1.2 DHCP & NTP Services

2.1 CUCM IP Phones

2.1a-CUCM: SCCP IP PHONE

2.1b-CUCM: SIP IP PHONE

2.2 CUCM Gateways

Information below is relevant to your digital T1 gateways.

 Line coding/framing b8zfs/esf

2.2a HQ CUCM: MGCP Gateway

R1# show ccm-mamanger host

2.2b-Site B CUCM: H.323 Gateway

2.3 CUCM SIP Trunks

2.3a-Inter-Site SIP Trunk using CUBE: HQ & Site B

2.3b-CUCM SIP Trunk to PSTN Troubleshooting

2.4 CUCM DIAL PLAN

General Dial-plan Rules:

Do not us the “@” wildcard in your CUCM dial plan.

2.4a-HQ PSTN Dial plan

 Configure local route group for HQ's local Voice gateway : R1

2.4b-Site B PSTN Dial plan

2.4c-SiteB Mobile Voice Access

Ensure the correct calling numbers are sent to all destinations.

2.4d-Site B Plus Dialing

2.4e-HQ SIP Trunk redundancy to PSTN

2.5 CUCM: URI Dialing

2.5a-HQ Intra-site URI Dialing

Create a URI for HQ Phone 1 (hqone@cciecollab.cisco.com) and HQ Phone 2

2.5b-HQ & SiteB Inter-site URI Dialing

2.6 CUCM: Media Resources

2.6a-HQ IOS Video Conference Bridge: R1

3.1 CUCME IP PHONES

3.1a-CUCME SCCP IP PHONES

3.1b-CUCME SIP IP Phones

3.1c-CUCME IP Phone Shared Lines

3.2 CUCME Gateway

Information below is relevant for your digital E1 gateway.

 Line code/framing HDB3/CRC4

3.2a-Site C E1 – PRI Trunk

3.3 CUCME PSTN Dial plan

3.3a-Site C PSTN Dial Plan

Site C PSTN Service Provider routing policies.

3.3b-Site C Inter-site Dial Plan to HQ & Site B

3.4 CUCME IP Phone Services and Media Resource

3.4a-Site C Hardware Ad-Hoc Conference Resources

3.4b-Site C IP Phone Customization

3.5 Cisco Unity Express

3.5a-Site C CUE Initialization

3.5b-CUE and CUCME Integration

Integrate CUE and CUCME using following information.

 Voice mail pilot number: 4220

4.1 QOS: Classification and Mapping

4.1a-Switch COS to DSCP Mapping

 Voice signaling CoS 3 to CS3

4.2 QOS: Congestion Management