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Lab 3: 01-APR-11

QUESTIONS Lab 3 WORKBOOK
Real Labs V3

ccievoicelabs.com | voice-labs.net

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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

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Section 3: Cisco Unified Communication Manager 3.1 CUCM IP Phones registration

Kindly Note :- 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, 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)

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3.2 CISCO CALL MANAGER EXPRESS
3.2.1 Customize phone background on CUCM. Images are kept in Candidate PC (142.100.64.16) customization of images has been already done.

Users should get selectable option for ccievoice image It should see in user preference and background image

Files are located on Candidate PC (142.102.64.16) on c: Voice-large.png Small-large.png

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3.2.2

Create a shared line 4003 between site C phone 1 and site C phone 2. 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. CME shared line, ensure 5 concurrent calls can be made into the DN. 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. 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

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Section 4: 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 circuit layer 2 should be user side. Calling names to be send to the PSTN Make inbound and outbound calls, Marks will not be given if calls won’t work.

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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 H323 T1-PRI gateway Configure CUCM to register SiteB Router controller T1 0/0/0 as IOS H323 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)

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4.3 Site C CME gateway Configure SiteC router as H323 gateway and register the same to CUCM. Use only 10 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

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Section 5: CUCM Call Routing PSTN access code for all IP phones– 9 Country code for US – 1 Country code for Hong Kong – 852 National code for HQ and SiteB IP phones – 1 International code for HQ and SiteB IP Phones – 011 International code for SiteC IP Phones – 00 5.1 CUCM Call Routing – HQ Gateway HQ PSTN provider specifications are as follows, 1) HQ PSTN provider expects proper information in “called party number” and “called party number type” fields. 2) “Called party number” and “called party number type” information must be set in ISDN setup messages. (Subscriber for local, National for long distance and International for International calls). 3) You MUST not use leading digit information to signal national (1) or international (011) calls. 4) If HQ Phone 1 makes international call to SiteC Phone 1 901185224044001, service provider expects “85224044001” in called party number field and “International” in “called party number type” field to route this call properly. 5) Unknown “Called party number type” field is only accepted for 911 emergency calls.

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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)

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5.2 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. 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 972303xxxx should be sent out to PSTN along with calling name. (3 points)

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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, 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. - +8522404xxxx. 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. (4 points)

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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 SB IP Phone 1 5252222 from SB PSTN phone 3033001. 2) On SB IP phone 1, it displays 7 digit calling number 5151111 along with calling name as “SB 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 +19725252222. 4) Select this call from list and click dial button to call this number. This should select SB gateway for call routing. 5) Once the call is connected it should show “TO 5252222”on SB Phone 1 display and “From 3033001” on PSTN Phone Display. This call should use SiteB gateway first. If SB gateway isn’t available then it should be routed via HQ gateway. When a call goes through HQ, caller ID should be 10 digits

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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 --------------- ----- --------------- ----- ------------ ----142.100.64.11 1720 142.100.64.11 32807 GK VOIP-GW H323-ID: GK-Trunk_1 Voice Capacity Max.= Avail.= Current.= 0 142.100.64.12 1720 142.100.64.12 32787 GK VOIP-GW H323-ID: GK-Trunk_2 Voice Capacity Max.= Avail.= Current.= 0 142.102.66.254 1720 142.102.66.254 65137 GK H323-GW H323-ID: CUCME Voice Capacity Max.= Avail.= Current.= 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: 142.102.66.254:1720 CUCME Prefix: 1* Zone GK master gateway list: 142.100.64.12:1720 GK-Trunk_2 142.100.64.11:1720 GK-Trunk_1

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You are not allowed to use default tech-prefix, zone subnet, and static alias commands. 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. Use 852 as tech-prefix to make calls to SiteC phones and 1 to make calls HQ phones from SC. 1) HQ/SB should be able to dial SC Phone by dialing 4 digit number & vice versa. 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

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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, the customer would like to send the calls via the backbone gatekeeper. Configure so that the calls to UK are sent via the backbone gatekeeper. Backbone Gatekeeper info: GK=BBGK Domain: cisco.com IP Address: 157.1.26.30 -connection HQ to an external GK is broken. -you have no access to the external GK. -List your troubleshooting work on a notepad file Write a report on the troubleshooting steps that you performed to accomplish this. Calls to +44 should be routed through the Backbone GK

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Section 6: Codec Selection Intra site calls should be G.711 and calls between sites should be G.729. Show gatekeeper calls, allocated bandwidth for each call should be 16kbps.

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Section 7: Media Resource Management 7.1 MOH When SiteB 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. 7.2 Call Park Call Park for HQ/SB with redundancy configured with null partition (range 2900 - 2902). 7.3 C-Barge CBarge should work on shared line for SiteC PH1 and SiteC PH2. (3 points)

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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
Ensure CoS 5 is mapped to EF 46. On port go 1/0/1 which is connected to HQ router, guarantee 16k for MGCP signaling traffic. Excess traffic should be marked to DSCP 8 and then transmitted.

8.2 Link fragmentation and Interleaving
There is a 384K link between HQ and SB and 768K between HQ and SC. Configure FRF.12 at a 10MS sampling rate there should be no header compression.

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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
Import HQPh1-HQPh3, SBPh1-SBPh2. You must import users from CUCM to achieve full marks. Use existing users in end users list. 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. Before leaving the lab MWI should be off. Test calls from HQ/SB to SC and vice versa. 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. (2 points)

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9.2 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 OR 142.1.64.254 OR 142.1.66.253

GUI web administrator : administrator GUI web password : ccievoice

9.3 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 . FTP Login details FTP Server IP : same candidate pc (access via VNC) FTP User name : administrator Pssword : ccievoice cue-vm-license_12mbx_cme_7.1.2.pkg cue-vm-langpack.nme.7.0.2.pkg cue-vm-k9.nme.7.1.2.pkg cue-vm-installer-k9.nme.7.1.2.prt1 cue-vm-en_US-langpack.nme.7.1.2.prt1 Note :(Already CTI port integrated and registered with CUCM . Once upload new license delete cti port configuration. Check the software lic file before proceed.

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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. Live Record Pilot 4250. CUE Live Record; make sure you are able to record a conference call by pressing live record softkey.

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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.

=======================================================
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”. If there are zero call in the queue, the script should play “Your Position is ‘X’. In other words let’s say if the first caller calls in, He/She should hear “Your Position is ZERO”. If the 2nd call comes in while the first call is in the queue, it should play “Your Position is ONE”.

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. Note: No agents need to be logged in. You don’t even need to configure an extension for IPCC. (5 Marks)

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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. When 4001 is off hook or in DND mode 3rd line of 4002 should be red.

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Section 12: High Availability 12.1 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. When such forwarded call comes to Cisco Unity connection, it should play user’s personal greeting. You are not allowed to use alternate extension to achieve this Make sure that the local, international and emergency calls work fine during SRST operation. 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).

12.2 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

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Call Forward Unregistered
If HQ or SC user calls the SB Phone and if it is not registered, he should be forwarded to SB Phone over the PSTN (For HQ to use HQ GW to call the Site B E.164 number. For SC to use the GK to call as an international number.) Provided a .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. When you call from HQ Phones calls should be routed through HQ Gateway. When you call from SC Phones calls should be routed through the GK and then HQ Gateway. Provided 2 screenshots of SiteB Phone 1:

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Forward (2001) For:+19723033001 (3...) By :+19723033001 (3...) Forward (4001) For:+19723033001 (3...) By :+19723033001 (3...)

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