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FINAL SET

Lab 3: 01-APR-11

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CCIE-VOICE-LABS.COM

VOICE-LABS.NET

FINAL SET

Lab 3: 01-APR-11

CCIE-VOICE-LABS.COM

VOICE-LABS.NET

FINAL SET

Lab 3: 01-APR-11

NET FINAL SET Lab 3: 01-APR-11 .CCIE-VOICE-LABS.COM VOICE-LABS.

NET FINAL SET Lab 3: 01-APR-11 .COM VOICE-LABS.CCIE-VOICE-LABS.

COM VOICE-LABS.NET FINAL SET Lab 3: 01-APR-11 .CCIE-VOICE-LABS.

NET FINAL SET Lab 3: 01-APR-11 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 .COM VOICE-LABS.CCIE-VOICE-LABS.

COM VOICE-LABS.NET FINAL SET Lab 3: 01-APR-11 .CCIE-VOICE-LABS.

CCIE-VOICE-LABS.COM VOICE-LABS.NET FINAL SET Lab 3: 01-APR-11 .

SiteB and SiteC to CUCM and assign extension numbers as specified in the above table. SiteC Phone 1 should display +85224044001. (3 points) .NET FINAL SET Lab 3: 01-APR-11 Section 3: Cisco Unified Communication Manager 3.1 CUCM IP Phones registration Kindly Note :. Extension-to-extension calling should use 4-digit dialing and should also deliver calling name.gHQ Phone 1 should display +14022022001.They change date-format and time to AM-PM you have to see in the lab and do it as per that! Register IP phones at HQ. 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.COM VOICE-LABS.CCIE-VOICE-LABS.

Images are kept in Candidate PC (142. Users should get selectable option for ccievoice image It should see in user preference and background image Files are located on Candidate PC (142.2 CISCO CALL MANAGER EXPRESS 3.64.COM VOICE-LABS.NET FINAL SET Lab 3: 01-APR-11 3.1 Customize phone background on CUCM.png .png Small-large.CCIE-VOICE-LABS.100.2.16) customization of images has been already done.102.16) on c: Voice-large.64.

COM VOICE-LABS. ensure 5 concurrent calls can be made into the DN. The phones should be able to barge in on an active call. Allow site C phone 1 and Phone 2 to make the call private when desired.2.CCIE-VOICE-LABS. CME shared line. OR * Shared line 4003 on SiteC PH1 and SiteC PH2 * Max 5 concurrent on shared DN 4003 * SiteC PH1 max incoming calls 4 * SiteC PH2 max incoming calls 2 * SiteC PH1 enable privacy button Configure a privacy button on 3rd line of phone 1 and Phone 2 .2 Create a shared line 4003 between site C phone 1 and site C phone 2. But STC phone 1 can only accept 2 inbound calls on this line at a time while STC phone 2 can accept 4 inbound concurrent calls.NET FINAL SET Lab 3: 01-APR-11 3.

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. Marks will not be given if calls won’t work. Calling names to be send to the PSTN Make inbound and outbound calls.NET FINAL SET Lab 3: 01-APR-11 Section 4: Voice Gateways and Signaling You will need the following information to complete the configuration. Your PRI circuit layer 2 should be user side. .COM VOICE-LABS.CCIE-VOICE-LABS.

(2 points) . Make sure that all inbound and outbound MGCP traffic is sourced from the local interface 142.254/24. Test the inbound calls to HQ IP Phones 408202xxxx where xxxx is extension range of HQ IP Phones.CCIE-VOICE-LABS.1 HQ IOS MGCP T1-PRI gateway Configure CUCM to register HQ Router controller T1 0/0/0 as IOS MGCP T1 PRI gateway.NET FINAL SET Lab 3: 01-APR-11 4. There is no need to test 9911 calling. Telco is sending 10-digits Direct-Inward-Dial (DID) for inbound PSTN calls.102.254/24. Calls made to this number should display 10-digit caller ID as 408202xxxx. Verify the gateway functionality by making outgoing calls to 911 emergency number. There is no need to test 9911 calling.COM VOICE-LABS. Telco is sending 10-digits Direct-Inward-Dial (DID) for inbound PSTN calls.64.2 SiteB IOS H323 T1-PRI gateway Configure CUCM to register SiteB Router controller T1 0/0/0 as IOS H323 T1 PRI gateway.102. Make sure that all inbound and outbound MGCP traffic is sourced from the local interface 142.65. Calls made to this number should display 10-digit caller ID as 972303xxxx. 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. (2 points) 4.

Telco is sending 8-digits Direct-Inward-Dial (DID) for inbound PSTN calls.254/24.3 Site C CME gateway Configure SiteC router as H323 gateway and register the same to CUCM.66.102. Calls made to this number should display 8-digit caller ID as 2404xxxx. Verify the gateway functionality by making outgoing calls to 999 emergency number.CCIE-VOICE-LABS. Make sure that all inbound and outbound H323 traffic is sourced from the local interface 142.NET FINAL SET Lab 3: 01-APR-11 4. (2 points) Note:POINTS WILL BE GIVING ONCE YOU WILL SUCCESSFULLY MAKE INBOUND AND OUTBOUND CALLS FROM 911/999 . Use only 10 channels of E1 PRI. Test the inbound calls to SiteC IP Phones 2404xxxx where xxxx is extension range of SiteC IP Phones.COM VOICE-LABS.

. 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. 3) You MUST not use leading digit information to signal national (1) or international (011) calls. 1) HQ PSTN provider expects proper information in “called party number” and “called party number type” fields.CCIE-VOICE-LABS. National for long distance and International for International calls).1 CUCM Call Routing – HQ Gateway HQ PSTN provider specifications are as follows. 5) Unknown “Called party number type” field is only accepted for 911 emergency calls.NET FINAL SET Lab 3: 01-APR-11 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. (Subscriber for local.COM VOICE-LABS. 2) “Called party number” and “called party number type” information must be set in ISDN setup messages.

3) Configure local route group for both the type of calls mentioned above so that it uses only HQ gateway for call routing. International calls should use only HQ gateway and no redundancy is required. Also. . Also.NET FINAL SET Lab 3: 01-APR-11 1) All HQ IP phones can make local PSTN calls by dialing 9 followed by 7 digit PSTN number. “called party number type” should be set to international for these calls.e. Second digit after the access code can be anything between 2 to 9. (3 points) . “called party number type” should be set to subscriber for these calls.+1408202xxxx. Calling number for such calls should be US country code leading “+” i.CCIE-VOICE-LABS. For such local calls.COM VOICE-LABS. Only HQ gateway should be selected and no redundancy is required. PSTN should send 7-digit calling number 202xxxx along with calling name. 2) All HQ IP phones can make International calls by dialing 9 followed by 011 then country code and variable length dialing digits. Rest of the digits can be anything between 0 to 9.

2) If SiteB IP Phone makes national call to numbers in 408 area code. 1) HQ PSTN provider uses leading digits in the called number to signal nonlocal calls.COM VOICE-LABS. 10-digit Calling number 972303xxxx should be sent out to PSTN along with calling name. PSTN should send 7-digit calling number 303xxxx along with calling name. 4) Unknown “Called party number type” field is only accepted for 911 emergency calls. if HQ gateway is not reachable. For such local calls. By considering the above specifications.NET FINAL SET Lab 3: 01-APR-11 5. HQ gateway should be selected to route these calls. 10-digit Calling number 972303xxxx should be sent out to PSTN along with calling name. service provider expects “01185224044001” in called party number field and to route this call properly. 2) “Called party number type” information can be ignored except local calls for which provider expects “subscriber” as “Called party number type” field. For above calls.2 CUCM Call Routing – SiteB Gateway SiteB PSTN provider specifications are as follows. (3 points) . it should use SiteB local gateway. Only SiteB gateway should be selected and no redundancy is required.CCIE-VOICE-LABS. 1 for national and 011 for international calls. 3) If SiteB Phone 1 makes international call to SiteC Phone 1 901185224044001. 1) All SiteB IP phones can make local PSTN calls by dialing 9 followed by 7 digit PSTN number. configure following requirements.

For such local calls.+8522404xxxx. Calling number for such calls should be Hong kong country code leading “+” i. 2) All SiteC IP phones can make International calls by dialing 9 followed by 00 then country code and variable length dialing digits. 3) If SiteC Phone 1 makes international call to HQ Phone 1 90014082022001. By considering the above specifications. International calls should use only SiteC gateway and no redundancy is required. 1) All SiteC IP phones can make local PSTN calls by dialing 9 followed by 8digit PSTN number. Also. “called party number type” should be set to international for these calls. (4 points) . configure following requirements.NET FINAL SET Lab 3: 01-APR-11 5. and International for International calls). (Subscriber for local. Also. 2) “Called party number” and “called party number type” information must be set in ISDN setup messages.3 CUCM Call Routing – SiteC Gateway SiteC PSTN provider specifications are as follows.COM VOICE-LABS.CCIE-VOICE-LABS. 1) SiteC PSTN provider expects proper information in “called party number” and “called party number type” fields. 4) Unknown “Called party number type” field is only accepted for 911 emergency calls. PSTN should send 8-digit calling number 2404xxxx along with calling name. “called party number type” should be set to subscriber for these calls. service provider expects “14082022001” in called party number field and “International” in “called party number type” field to route this call properly.e. . Only SiteC gateway should be selected and no redundancy is required.

3) Press directories button to go to missed call menu. Use “debug isdn q931” output to verify number type information for calling and called number sent by PSTN. After selecting missed calls menu.COM VOICE-LABS. caller ID should be 10 digits . This should select SB gateway for call routing. it displays 7 digit calling number 5151111 along with calling name as “SB PSTN”. This call should use SiteB gateway first. If SB gateway isn’t available then it should be routed via HQ gateway. this call should display globalized calling number +19725252222.4 CUCM Call Routing – “+” dialing consideration Configure CUCM to deliver globalized dialing pattern for HQ IP phones.CCIE-VOICE-LABS. 4) Select this call from list and click dial button to call this number. 5) Once the call is connected it should show “TO 5252222”on SB Phone 1 display and “From 3033001” on PSTN Phone Display. Do not answer this call.NET FINAL SET Lab 3: 01-APR-11 5. Refer to below example. When a call goes through HQ. 1) Make inbound call to SB IP Phone 1 5252222 from SB PSTN phone 3033001. 2) On SB IP phone 1.

66.100.100.COM VOICE-LABS.64.-----------.----.254 1720 142.= Current.100.102.64.NET FINAL SET Lab 3: 01-APR-11 Gatekeeper Call Routing Register the CUCM and CME to match the following outputs that are in bold: HQ-RTR# sh gatekeeper endpoints GATEKEEPER ENDPOINT REGISTRATION ================================ CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags --------------.CCIE-VOICE-LABS.= Avail.12:1720 GK-Trunk_2 142.= Avail.11 32807 GK VOIP-GW H323-ID: GK-Trunk_1 Voice Capacity Max.102.--------------.= Current.254:1720 CUCME Prefix: 1* Zone GK master gateway list: 142.----.100.64.= Current.102.66.100.11:1720 GK-Trunk_1 .64.= 0 Total number of active registrations = 3 HQ-RTR# sh gatekeeper gw-type-prefix GATEWAY TYPE PREFIX TABLE ========================= Prefix: 852* Zone GK master gateway list: 142.= 0 142.12 1720 142.12 32787 GK VOIP-GW H323-ID: GK-Trunk_2 Voice Capacity Max.66.100.= Avail.64.64.= 0 142.254 65137 GK H323-GW H323-ID: CUCME Voice Capacity Max.11 1720 142.----142.

SiteC should use its vlan address for all communications with the gatekeeper HQ phones should be able to call SiteC phones by dialing 4 digits internal extensions.NET FINAL SET Lab 3: 01-APR-11 You are not allowed to use default tech-prefix. zone subnet. Use 852 as tech-prefix to make calls to SiteC phones and 1 to make calls HQ phones from SC.CCIE-VOICE-LABS. 1) HQ/SB should be able to dial SC Phone by dialing 4 digit number & vice versa.COM VOICE-LABS. and static alias commands. 2) If in any case if gatekeeper is down calls should be routed from the backup path and reach to the destination and in this case calling id should be E164 .

-you have no access to the external GK. Backbone Gatekeeper info: GK=BBGK Domain: cisco. Configure so that the calls to UK are sent via the backbone gatekeeper.NET FINAL SET Lab 3: 01-APR-11 Gatekeeper troubleshooting section We have a customer that places high call volume to UK resulting in high cost. In order to avoid high toll charges with these calls.30 -connection HQ to an external GK is broken.com IP Address: 157.COM VOICE-LABS. the customer would like to send the calls via the backbone gatekeeper. -List your troubleshooting work on a notepad file Write a report on the troubleshooting steps that you performed to accomplish this.1.CCIE-VOICE-LABS.26. Calls to +44 should be routed through the Backbone GK .

COM VOICE-LABS.CCIE-VOICE-LABS. Show gatekeeper calls.729. .NET FINAL SET Lab 3: 01-APR-11 Section 6: Codec Selection Intra site calls should be G. allocated bandwidth for each call should be 16kbps.711 and calls between sites should be G.

7. configure local routers to stream G711 multicast MOH from router flash.3 C-Barge CBarge should work on shared line for SiteC PH1 and SiteC PH2.au” file in router flash for this multicast requirement.1 MOH When SiteB IP phones or PSTN users are put on hold.2902). You can use “music-onhold.NET FINAL SET Lab 3: 01-APR-11 Section 7: Media Resource Management 7.CCIE-VOICE-LABS. (3 points) . 7.COM VOICE-LABS.2 Call Park Call Park for HQ/SB with redundancy configured with null partition (range 2900 .

Configure FRF.1 Switch QoS Ensure CoS 5 is mapped to EF 46. 8. . guarantee 16k for MGCP signaling traffic. Excess traffic should be marked to DSCP 8 and then transmitted. 8. If there is any such impact.12 at a 10MS sampling rate there should be no header compression.COM VOICE-LABS.2 Link fragmentation and Interleaving There is a 384K link between HQ and SB and 768K between HQ and SC. this section will not be marked.NET FINAL SET Lab 3: 01-APR-11 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. On port go 1/0/1 which is connected to HQ router.CCIE-VOICE-LABS.

You must import users from CUCM to achieve full marks. Before leaving the lab MWI should be off. make sure that voicemail pilot numbers for both Cisco unity Connection as well as Cisco unity express are reachable from PSTN.1 Cisco Unity Connection Integration and Configuration Cisco Unity Connection is pre-configured and integrated with CUCM with following Configuration. Set user passwords to 12345 Pilot Number for voice mailbox should be reachable from PSTN Make sure CUC/CUE voicemail greetings and MWI should work. Make sure to clear MWI once you test the same in the lab. For HQ Phone1 make sure if PSTN caller left a voicemail the user can hear the calling number of the PSTN caller and the message disposition time before playback the message. Voicemail Pilot – 2220 Voicemail ports – 2221-24 MWI On – 1998 MWI off – 1999 AXL username – administrator AXL password – ccievoice Import HQPh1-HQPh3. 9. Test calls from HQ/SB to SC and vice versa. Use existing users in end users list. (2 points) . SBPh1-SBPh2.NET FINAL SET Lab 3: 01-APR-11 Section 9: Voice Mail Integration You should check MWI functionality for Cisco Unity connection as well as Cisco Unity Express.COM VOICE-LABS. Also.CCIE-VOICE-LABS.

prt1 cue-vm-en_US-langpack. IP Address : 142. FTP Login details FTP Server IP : same candidate pc (access via VNC) FTP User name : administrator Pssword : ccievoice cue-vm-license_12mbx_cme_7. Check the software lic file before proceed.253 Hostname : CUE Domain name : ccievoice.66.7.0.COM VOICE-LABS.102.pkg cue-vm-langpack.pkg cue-vm-k9.2.nme.CCIE-VOICE-LABS.nme. Once upload new license delete cti port configuration.2.253 GUI web administrator : administrator GUI web password : ccievoice 9.1.3 Cisco Unity Express configuration and CUCME integration Change CUE license file to CUCME and integrate the same with CUCME Following license files available FTP server .1.com DNS : not required NTP : 142.254 OR 142.NET FINAL SET Lab 3: 01-APR-11 9. You need to run through the initial setup wizard to configure following settings.66.nme.pkg cue-vm-installer-k9.64.2 Cisco Unity Express Initial Configuration Cisco Unity Express is set to factory default settings.102.1.7.2. .64.7.254 OR 142.2.7.1.nme.1.1.prt1 Note :(Already CTI port integrated and registered with CUCM .2.

.NET FINAL SET Lab 3: 01-APR-11 9.4 Advanced CUE Users from HQ and SB should be able to reach CUE voicemail and it should succeed Configure Live Record for SiteC users. make sure you are able to record a conference call by pressing live record softkey.CCIE-VOICE-LABS. Live Record Pilot 4250.COM VOICE-LABS. CUE Live Record.

In other words let’s say if the first caller calls in.CCIE-VOICE-LABS. You don’t even need to configure an extension for IPCC. it should play “Your Position is ONE”. If there are zero call in the queue. ======================================================= Create an script in such a way so that when users call in they hear “Thank you for calling” and immediately after that it should play “All of our representatives are busy at this time please stay on the line someone will be with you shortly”. (5 Marks) . If the 2nd call comes in while the first call is in the queue. First call "your position in the queue is 0" / second call "your position in the queue is 1" You are asked by your customer to generate the necessary prompts to fulfill the above mentioned requirements by using the UC voice recording tools available on your POD. He/She should hear “Your Position is ZERO”.COM VOICE-LABS. Note: No agents need to be logged in.NET FINAL SET Lab 3: 01-APR-11 Section 10: UCCX Applications EVENT Tracker Message Event trackers message comes when you use the RDP/Remote desktop you can put any reason and start the same. the script should play “Your Position is ‘X’.

NET FINAL SET Lab 3: 01-APR-11 CME Presence SiteC PH2 button 3 should monitor the status of SiteC PH1 primary line when off-hook and DND status. This should function also as a speed dial button to call SiteC PH1 Monitor line status of 4001 from 3rd line of 4002. .CCIE-VOICE-LABS.COM VOICE-LABS. When 4001 is off hook or in DND mode 3rd line of 4002 should be red.

NET FINAL SET Lab 3: 01-APR-11 Section 12: High Availability 12. When such forwarded call comes to Cisco Unity connection. 911 (send 10 digits callerid) local (send 7 digits callerid) International (send callerid e164) Make sure 4 digit call should work between SB-HQ & SB-SC during WAN failure (send callerid e164).CCIE-VOICE-LABS. 12. international and emergency calls work fine during SRST operation.1 Site B router high availability You cannot use CME SRST. you must configure Call-Manager-FallBack 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. it should play user’s personal greeting.COM VOICE-LABS.2 SRST Advance Voice Mail should work in case of busy or incoming call will be ringing for 20 secs Make sure all the incoming and outgoing call should same way as it is registered with CUCM Phone should work same as it is registered with CUCM . You are not allowed to use alternate extension to achieve this Make sure that the local.

NET FINAL SET Lab 3: 01-APR-11 Call Forward Unregistered If HQ or SC user calls the SB Phone and if it is not registered.COM VOICE-LABS. he should be forwarded to SB Phone over the PSTN (For HQ to use HQ GW to call the Site B E. When you call from SC Phones calls should be routed through the GK and then HQ Gateway. When you call from HQ Phones calls should be routed through HQ Gateway. For SC to use the GK to call as an international number.) Provided a .CCIE-VOICE-LABS. Provided 2 screenshots of SiteB Phone 1: .164 number.2screenshot of a phone at siteC and the phone should display: Forwarding from: +19723033001 OR Make sure that HQ/SC Phones are be able to call SB PH1 using 4 digit dialing in event of a WAN failure.

.) By :+19723033001 (3.CCIE-VOICE-LABS....NET FINAL SET Lab 3: 01-APR-11 Forward (2001) For:+19723033001 (3.) .COM VOICE-LABS.) Forward (4001) For:+19723033001 (3...) By :+19723033001 (3...

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