Server CUCM Pub CUCM Sub Cisco Unity Connection UCCX CUPS CUPC Test machine

IP Address 142.100.64.11 142.100.64.12 142.100.64.13 142.100.64.14 142.100.64.15 142.100.64.16

Site HQ

VLAN Server Voice Data

VLAN ID 100 102 202 302 402 502 602

IP Address 142.100.64.0/24 142.102.64.0/24 142.202.64.0/24 142.102.65.0/24 142.202.65.0/24 142.102.66.0/24 142.202.66.0/24

SiteB

Voice Data

SiteC

Voice Data

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

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

Phone HQ Phone 1 HQ Phone 2 HQ Phone 3 SiteB Phone 1 SiteB Phone 2 SiteC Phone 1 SiteC Phone 2

PSTN Numbering Plan 1 408 202 2001 1 408 202 2002 1 408 202 2003 1 972 303 3001 1 972 303 3002 852 2404 4001 852 2404 4001

Phone HQ Phone 1 HQ Phone 2 HQ Phone 3 SiteB Phone 1 SiteB Phone 2 SiteC Phone 1 SiteC Phone 2

Internal Numbering Plan 2001 2002 2003 3001 3002 4001 4002

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

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

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

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

See another file for the same

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.gHQ 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 900185224044001, 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 nonlocal 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 900185224044001, service provider expects 00185224044001 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 8digit 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-onhold.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 6th 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 6th 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)