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Questions lab-4 Guide

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! Also you have to see that this image they give for SB phones so you have to modify 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 & CUCM Create a share line between HQPH1 & HQPH2 number is 2012 Create share line HQPH1, max calls 5, and inbound 4 Create share line HQPH2, max calls 5, and inbound 3

(3 Points)

Privacy Button on 3rd line of HQ1 and HQ2. Make sure when this button is pressed, the other phone cannot see the calling number of the shared line. Privacy should work on all the states (3 Points)

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INTERCOM Configure Intercom on SCPH1 & SCPH2 Configure Intercom lines on button 2 of Phone 1 and 2 (could be any extension) Intercom DNs shouldnt be dial able from any other extension. When there is an active call on SC Ph2 and the intercom call arrives from SC Ph1 It should put the active call on hold and force the intercom line to become active. When the call is active on SCPH1 and the intercom call arrives from SCPH2, it should ring on the intercom line instead of going to auto-answer.

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Background image on SCPH1 & SCPH2

(4 Points)

Images are kept in Candidate PC (142.100.64.16) customization of images has been already done. Users should get image on ip phone 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 Images are found on the candidate test pc in the candidate folder on the desktop. Images are call voice-large and voice-small. Note. No TFTP server is provided. You are required to upload these to CUCM and download to CUCME. This is not mentioned in the lab.

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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 wont 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 CUCME. 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 MGCP 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,

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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 H323 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 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)

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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, 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. - CME Call Routing SiteC Gateway - Local calls (send 8 digits callerid) - International calls (send callerid e164) (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 5252222on SB Phone 1 display and From 3033001 on PSTN Phone Display. This call should use SiteB gateway first. If SB gateway isnt available then it should be routed via HQ gateway. When a call goes through HQ, caller ID should be 10 digits This is ccievoicelabs . com labs this statement is written so that no one can steal the lab

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MGCP Troubleshooting 1) MGCP -- R1 Register with MGCP. -- Test this gateway by making 911 calls to the pstn phone. Also make inbound calls from Line 1 on the pstn phone (4085151111) by dial 202XXXX. -- Use the sub as the primary call agent and the pub as the secondary -- When the mgcp is registered to the secondary call-agent when the primary call-agent comes back up it should register immediacy with the primary call-agent (3 Points)

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MGCP TROUBLESHOOTING KINDLY NOTE :There is 2 variations for MGCP Troubleshooting please see the First Event 2) (4 Points)

Management wants you to prove when subscriber fail publisher will work Call 911 while the call is active shut down the cucm sub services. Make sure when Complete to bring up the subscriber services so you can continue with the tasks on the lab. Capture the following -- Backup call agent sends the message to the gateway to check the status of the active call OR -- Backup call agent sends the message to the gateway to check the status of the First call -- Gateway sends the status of active calls to the secondary call agent. -- Back up call agent send a message back to the GW requesting additional information about the call. Put all the steps as Event 1, Event 2 and Event 3 in the notepad and save it in the candidate test pc as mgcp.txt.

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Gatekeeper Section You are not allowed to use default tech-prefix, zone subnet, and static alias commands. SiteC should use its loopback 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

-- HQ & SB both can call SC 400X number through the gatekeeper with 4 digit ANI. If the gatekeeper is down HQ and SB both should route over the PSTN and display the fully globalized number -- SC can call HQ and SB across the gatekeeper and should display 4 digits as ANI. If the gatekeeper is down SC can call HQ and SB across the pstn and should display the fully globalized number +8522404400X. Note - Use isdn called numbering type of international for back up scenario

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Gatekeeper Advance section CUCM users calling Belgium phone number with the country code 32 this call is sent to GK Gk matches the 01132 with the INT code 011 this call is sent to BBGK BBGK details Zone : BBGK Domain : cisco.com IP : 157.26.100.253 Security purpose the HQ phone calls should be routed to the BBGK with the IP 142.102.64.254 In other words the RTP voice stream has to go through the GK to BBGK. This should be restricted from endpoint to endpoint Dont use MTP or transcoder to achieve it. Sent G729 calls to the BBGK.

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Gatekeeper troubleshooting section

HQ phones are complaining that when the Belgium call is connected they get ring back tone. On the other hand Belgium user x get no voice and they drop the call. Please do the troubleshooting and put it in the notepad Belgium.txt.

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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. This is ccievoicelabs . com labs this statement is written so that no one can steal the lab

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Section 7: Media Resource Management Cbarge Create a shared line in Site B ph 1 and ph 2 3012. Make sure it only uses a hardware conference bridge. OR Configure c-barge but do not use the softkey template for siteb (3 points)

Call Park HQ ph 2 can transfer the call to the call park no 2802. HQ phone 2 can retrieve the call by calling *2802.
Extra If the call is not retrieved it has to try to redeliver in 45 secs. This feature should be available only for HQ ph2.

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Meet me conference -

(4 points)

-- SITEC Ph can initiate the meet me conference the other users can call the meet me number and get connected to the conference. PSTN can also access the conference bridge 4321 is the number for the meet me. -- Make sure when user join and leave the conference beeps are heard

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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 LAN QOS - 5 points

COS COS COS COS 5 should be in priority queue 4, 6, 7 should be in Queue 2 3, 2, 3 should be in Queue 3 4, 0 should be in Queue 4

Guarantee Queue 1 has the 25% of the bandwidth the other queues should share the bandwidth as 30 40 30. Once queue 2 reaches 60% capacity COS 4 packets should be dropped.

8.2 Link fragmentation and Interleaving

There is a 384K link between HQ and STB and 768K between HQ and STC. Configure FRF.12 with interleaving and fragmentation delay of 10ms HQ -- SB (FRF.12) HQ -- SC (FRF.12)

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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. Use existing users in end users list. Set user passwords to 12345 Pilot Number for voice is reachable from PSTN Make sure CUC/CUE voicemail greetings and MWI work. Test calls from HQ/SB to SC and vice versa. Calls should go to voicemail after 20 seconds or if the caller is on the line Verify MWI works but make sure MWI is not on when you leave the lab (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 : 142.1.66.253 Hostname : CUE Domain name : ccievoice.com DNS : not required NTP : 142.1.64.254 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. OR Note:Once Initialized verify the license file.

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EXTRA QUESTION Install GB language package in CUE

-- Set up a SC ph1 and ph2 user for cue and set their password to 12345 -- Verify MWI works but make sure WMI is not on when you leave the lab -- Verify that calls placed from the PSTN as well as call from HQ and SB phones across the gatekeeper can leave a message for a user at SC

9.4 Advanced CUE Users would like it when they listen to a message from a pstn caller the are able to hear the calling number prior to the message in the users envelope.

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Section 10: UCCX Applications Create the following script

(5 points)

-- Create script in such a way so that when users call they should 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 some one will be with you shortly. -- If there are zero call in the queue, the script should play There are currently X calls ahead of you. -- In other words lets say if the first caller calls in queue, He/She should hear There are currently ZERO calls ahead of you. If the 2nd call comes in while the first call is in the queue, it should play There are currently ONE calls ahead of you. -- 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 dont even need to configure an extension for IPCC also no need for calls from SB to be routed to UCCX. Only pstn and HQ callers. (5 Marks)

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CME Presence

(5 points)

-- SCPh2 should be able to monitor SCph1 line 1 their should be a 3rd line on SCPh2 that monitors this phone. When you push this button it should speed dial to 4001. When 4001 is on the phone this button should show red -- When phone 1 line 1 (4001) is on the phone you should see the status of this call in the local directory of phone 1 as shown in the picture.

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Section 12: High Availability 12.1 Site B router high availability 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 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 calls should work between SB-HQ & SB-SC during WAN failure (Send callerid e164).

SRST ADVANCE

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-- While in SRST phones should appear exactly like when they are registered to CUCM except for the message "Your phones are in fallback" displayed at the bottom of the phones -- Both of the primary lines on each phone sbph1 and sbph2 should be able to make or receive more than two calls -- Call forword no answer should work in SRST for SB Phone 1 -- Your are not allowed to have information for learned ephones in the running configuration -- When in SRST SB call should be able to call hq phones and should display 4 digit ANI as caller id. Assume the Telco can understand this. Note not mention of HQ calling SB while in SRST.

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12.2 CUCM Call forward unregistered Make sure that all HQ IP phones should be able to call 3001 using 4-digit dialing in event of WAN failure. (2 points)