Lab 2 Ccievoicelabs Reallabs
Short Description
Lab 2 Ccievoicelabs Reallabs...
Description
CCIE-VOICE-LABS.COM VOICE-LABS.NET
Lab 2: 01-OCT-10
CCIE-VOICE-LABS.COM VOICE-LABS.NET
Lab 2: 01-OCT-10
LOGICAL OVERVIEW DIAGRAM CCIEVOICELABS
CCM_Pub 64.11
UNITY_CONN 64.13
CCM_Sub 64.12
HQ Server Vlan 100 HQ Voice Vlan 102 HQ Data Vlan 202 1 408-202-2XXX
UCCX 64.14
CUPS_Test_PC 64.15
CUPS 64.15
CC
HQ Site 3750 Switch
MGCP 2001 (7965)
2002 (7965)
2003 (7965)
HQ Gateway (PST) 64.254
3825
PSTN Number: 911 999 1 408-202-2222 1 972-230-3333 852-2404444
T1 - PRI
1500 K (?)
PSTN Phone 384 K
MGCP SB Voice Vlan 302 SB Data Vlan 402 1- 972-303-3XXX
28XX
E1 - PRI T1 - PRI Site B (EST) 65.254
EtherSwitch
3001 (7965)
1200 K
Site C (HK) 66.254
H323 GW (CUE)
28XX
EtherSwitch
3002 (7965)
4001 (7965)
SC Voice Vlan 502 SC Data Vlan 602 852-2404-4XXX
4002 (7965)
CCIE-VOICE-LABS.COM VOICE-LABS.NET
Lab 2: 01-OCT-10
CCIE-VOICE-LABS.COM VOICE-LABS.NET
Lab 2: 01-OCT-10
CCIE-VOICE-LABS.COM VOICE-LABS.NET
Lab 2: 01-OCT-10
CCIE-VOICE-LABS.COM VOICE-LABS.NET
USERID HQ PH 1 HQ PH 2 SB PH 1 SB PH 2 SB PH 3 Uccxadmin ProctoX
PIN 12345 12345 12345 12345 12345 ccievoice ccievoice
User id are already create and do not delete or modify the same
Lab 2: 01-OCT-10
CCIE-VOICE-LABS.COM VOICE-LABS.NET
Lab 2: 01-OCT-10
Section 2: Basic Campus Design 1.1 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)
CCIE-VOICE-LABS.COM VOICE-LABS.NET
Lab 2: 01-OCT-10
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)
CCIE-VOICE-LABS.COM VOICE-LABS.NET
Lab 2: 01-OCT-10
Section 2: Cisco Unified Communication Manager 2.1 CUCM IP Phones registration
Same as SC just they change date to (–) & time to AM or PM 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)
CCIE-VOICE-LABS.COM VOICE-LABS.NET
Lab 2: 01-OCT-10
2.2 IP Phone customization (Part I) HQ phone 1 should see the DND on the phone screen. Once the requirement has been met. Do the following. • DND softkey should be there in idle and active conversation state. •
DND softkey should be available while incoming calls ringing.
•
When DND is pressed incoming call should directly go to Voicemail.
•
When DND is depressed and the phone rings still it should go to Voicemail.
2.2 IP Phone customization (Part II)
Configure Intercom on phone HQ Phone 1 and HQ Phone 2 on 6th Line. 2021 and 2022 respectively. “PHOTO of Intercom Phone” • Intercom user should invoke and seize by pressing the intercom line and dial the target phone, Target phone should automatically get answered with whisper (muted enabled) Target phone can make two way audio able by pressing the intercom button.
CCIE-VOICE-LABS.COM VOICE-LABS.NET
Lab 2: 01-OCT-10
2.2 IP Phone customization (Part III) Users should get selectable option for ccievoice image
Customize phone background on CUCM. Files are located on Candidate PC (142.102.64.16) on c: Voice-large.png Small-large.png
CCIE-VOICE-LABS.COM VOICE-LABS.NET
Lab 2: 01-OCT-10
Section 3: Voice Gateways and Signaling You will need the following information to complete the configuration. For the T1 controller: Switch Type: primary-ni Framing 8BZS Line Code: ESF For the E1 controller: Switch Type: primary-net5 Framing CRC4 Line Code: HDB3 Take clocking for Layer 1 from Network side. Your PRI clocking of layer 2 should be user side.
3.1 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)
CCIE-VOICE-LABS.COM VOICE-LABS.NET
Lab 2: 01-OCT-10
3.2 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)
3.3 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) Note:POINTS WILL BE GIVING ONCE YOU WILL SUCCESSFULLY MAKE INBOUND AND OUTBOUND CALLS FROM 911/999
CCIE-VOICE-LABS.COM VOICE-LABS.NET
Lab 2: 01-OCT-10
Section 4: 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 4.1 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.
CCIE-VOICE-LABS.COM VOICE-LABS.NET
Lab 2: 01-OCT-10
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) If HQ IP Phone makes national call to numbers in 972 area code, SiteB Gateway should be selected first to route these calls. 10-digit Calling number 408202xxxx should be sent out to PSTN along with calling name. If SiteB gateway is not available, it should use local HQ gateway to route these calls. 3) 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. 4) Configure local route group for both the type of calls mentioned above so that it uses only HQ gateway for call routing. (4 points)
4.2 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 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.
CCIE-VOICE-LABS.COM VOICE-LABS.NET
Lab 2: 01-OCT-10
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 303xxxx 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 972303xxxx should be sent out to PSTN along with calling name. 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. (4 points) 4.3 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,
CCIE-VOICE-LABS.COM VOICE-LABS.NET
Lab 2: 01-OCT-10
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. - +8522404xxxx. And for example if you want to dial the US number you have to dial 0014082022001 International calls should use only SiteC gateway and if you SiteC gateway is not available then call should go from SiteB gateway. Some users do not like to wait for E164 timer so also create a patter with # for fast dialing. Also, “called party number type” should be set to international for these calls. 3) If SiteC dial the international number to HQ phones it should take IP WAN rather then going from PSTN network so when SiteC user will call to HQ Phone 1 it should display SiteC phone 1 on HQ phone. 4) International calls from SiteC IP Phones to HQ numbers in the range of 1408XXXXXXX should be routed via HQ gateway first as local calls. If HQ gateway is unavailable, it should use local SiteC gateway. Calling number for such calls should be Hong kong country code leading “+” i.e. - +18522404xxxx. 5) Configure local route group for both the type of calls mentioned above so that it uses only SiteC gateway for call routing. (4 points)
CCIE-VOICE-LABS.COM VOICE-LABS.NET
Lab 2: 01-OCT-10
5.1 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 5151111. 2) On HQ IP phone 1, it displays 7 digit calling number 5151111 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 +14085151111. 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 5151111” on HQ IP phone 1 display and “From 2022001” on PSTN phone display. (3 points)
CCIE-VOICE-LABS.COM VOICE-LABS.NET
Lab 2: 01-OCT-10
5.2 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 25353333. 2) On HQ IP phone 1, it displays calling number as 85225353333 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 +85225353333. 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 25353333” 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)
CCIE-VOICE-LABS.COM VOICE-LABS.NET
Lab 2: 01-OCT-10
5.4 Single Number Reach – Cisco Mobile Connect Mobility softkey should be on display of HQ Phone 3 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 5251111 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 3, 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) When user has taken the call from PSTN phone he should be able to re-route to IP phone by holding in 30 sec (3 points)
CCIE-VOICE-LABS.COM VOICE-LABS.NET
Lab 2: 01-OCT-10
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)
CCIE-VOICE-LABS.COM VOICE-LABS.NET
Lab 2: 01-OCT-10
Section 7: Media Resource Management 7.1 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) 7.3 Call Park * Configure Call Park for Site C users only. * Call Park Range should be 4300 - 4302. * Call park should work in cucm failover state. * Call park numeric and wildcard should be same to achieve this task.
CCIE-VOICE-LABS.COM VOICE-LABS.NET
Lab 2: 01-OCT-10
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 LAN QOS - 5 points COS COS COS COS
5 should be in priority queue 1, 3, 6 should be in Queue 2 2, 4 should be in Queue 3 0 should be in Queue 4
On Gi1/0/1 Guarantee traffic for priority Queue. Queue 2 , Queue 3 and Queue4 should share the bandwidth as 20%, 40% , 40%. Queue 1 should get one third of BW on Link
OR
Map “COS 5” to DSCP value of EF 2) On port fa 1/0/13 which is connected to HQ Phone 1, guarantee 8k 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.
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)
CCIE-VOICE-LABS.COM VOICE-LABS.NET
Lab 2: 01-OCT-10
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)
CCIE-VOICE-LABS.COM VOICE-LABS.NET
Lab 2: 01-OCT-10
9.2 Unity Subscriber Customization When someone calls from PSTN it should not say HQ PHONE 1 not available it should say 2001 is not available please leave your message after the tone. You are not allowed to modify the personal greetings to achieve this task. Also you have to import the users from CUCM to Unity to get full marks.
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)
CCIE-VOICE-LABS.COM VOICE-LABS.NET
Lab 2: 01-OCT-10
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)
CCIE-VOICE-LABS.COM VOICE-LABS.NET
Lab 2: 01-OCT-10
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 When someone calls on business hours (8 to 5) Monday to Friday it should play a greeting “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”. “You can modify the ICD script to achieve this task” But after hours it should play “Please try again later” You do not need to record the prompt No agents need to be logged-in. “You have to use the system prompt to achieve this task” (3 points)
CCIE-VOICE-LABS.COM VOICE-LABS.NET
Lab 2: 01-OCT-10
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. If someone calls on HQ Phone 1 and it is in ringing mode HQ Phone 3 should show the Orange LED blinking. (2 points)
CCIE-VOICE-LABS.COM VOICE-LABS.NET
Lab 2: 01-OCT-10
11.2 Cisco Unified Presence server and client Integrate Cisco Unified Presence server with CUCM to achieve following requirement, 1) Configure presence to control SB PH 1. 2) Do not enable Desktop Phone Configuration on this presence client. 3) Enable Desktop control. Integrate Cisco Unified Presence server with CUCM to achieve following requirements, - Call control only for SB Ph 1 3001 should be available - Change Proxy Domain in Service Parameter to ccievoice.com - Add cups & cups.ccievoice.com to Host file of Test PC Windows XP (142.100.64.16)
CCIE-VOICE-LABS.COM VOICE-LABS.NET
Lab 2: 01-OCT-10
Section 12: High Availability 12.1 SiteC router 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, long distance and international calls. Such calls made should display 10-digit caller ID. 2) Make sure that CUE 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 CUE, it should play user’s personal greeting. MWI should work in SRST mode for SC users. 12.2 SRST Advance Call park should work in SRST mode.
View more...
Comments
