Cisco VoiceIE资料

1 Lab Topology :1.1 Lab Dial Plan and Addressing:!Tip -All Sub-interface and L3 VLAN interface for server, voice and data are assigned ip address .254 from the respective subnet.!Tip –R2 and R3 routers have HWIC-4ESW for connecting IP Phone.2 Basic Campus Design2.1Voice and Data VLANs (2 points)Configure Voice VLANs for switch ports connecting to IP Phones at HQ, SiteBandSiteC.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 placedin 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.SB SC 4ESWRack06R1vlan 100 en 6 int f0/0.100 f0/0.6 server interface ip 142.100.64.254/24vlan 106 en 106 int f0/0.106 hq voice 142.102.64.254vlan 206 en 206 int f0/0.206 hq data 142.202.64.254in loo 0 142.1.64.254/32 24int s02/0R1-R2 s0/2/0.102 p to p 156.106.26.1/24 dlci 102R1-R3 s0/2/0.103 p to p 156.206.26.1/24 dlci 103R1-TO-RACK0506BB如果用BBint f0/0.156 en 156 157.26.1.254不用BBint loo 157interface FastEthernet1/0/2description To-Rack03R1-Phone switchport access vlan 203switchport mode accessswitchport voice vlan 103spanning-treeportfastR2vlan 306 en 306 int f0/0.306 sb voice vlan 406 en 406 int f0/0.406 sb dataint loo 0 142.1.65.254/324eswvlan databasevlan 306 name sb-voicevlan 406 name sb-dataintvlan 306ip add 142.102.65.254interface FastEthernet0/3/3switchport trunk native vlan 403 switchport mode trunkswitchport voice vlan 303interface FastEthernet0/3/3switchport trunk native vlan 403 switchport mode trunkswitchport voice vlan 303s0/2/0R2-R1int s0/2/0.201 p to p 156.106.26.2 dlci 201 R3vlan 506 en 506 int f0/0.506 sc voice vlan 606 en 606 int f0/0.606 sc dateint loo 0 142.1.66.254/32s0/2/0R3-R1int s0/2/0.301 p to p 156.206.26.2 dlci 301 noteR1 R2 R31. R1#show vlan-switch2. show run int s0/2/0.102R1(config-subif)#ippim sparse-dense-mode3 R2 R3 4ESWswitch 37501. check interface to phone vlan information2. cdp advertise-v23. interface to ccm, duplex half2.2DHCP Service (2 points)Configure CUCM Publisher as DHCP server to provide lP Addresses for lPPhones at HQ and SiteB from their respective Voice subnets.For HQ, use lP address range from 142.102.64.10/24 to 142.102.64.30/24 For SiteB, use lP address rangefrom 142.102.65.10/24 to 142.102.65.30/24HQ Option 150 ip 142.100.64.12 142.100.64.11Configure local Cisco 2811 router as DHCP server to provide IP addresses for SiteClP Phones from localVoice subnet.Use IP address range from 142.102.66.10/24 to 142.102.66.30/24 SC Option 150 ip 142.102.66.254NoteCCM PUB DHCP SERVER 142.100.64.11hq voice subnet142.102.64.10 142.102.64.30gw 142.102.64.254option 150 142.100.64.12 142.100.64.11sb voice142.102.65.10 142.102.65.30gw 142.102.65.254option 150 142.100.64.12 142.100.64.11sc voice142.102.66.10 142.102.65.30gw 142.102.66.254option 150 142.02.66.254CCM inite1. check service2. rename ccm for name to ip3.Cisco Unified CM Group Configuration pub behind sub4.Enterprise Parameters Configurationnote1. gateway2. ip helper3. dhcp service .restart dhcp service4. inite all phone2.3 NTP (2 points)HQ PST -8SB CST -6SC CCT +8CCM ,UNITYCN,Presence ServerSynchronize HQ router with external NTP source at 157.26.1.253. 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 theybelong.Phones should display 12hours time format. Date displayed should be in the format mm-dd-yy.hq route NTP server 157.26.1.253sb route ntp server 142.1.64.254sc route ntp server 142.1.64.254CCM PUB ntp server 142.1.64.254CCM SUB ntp server 142.100.64.11 CCM pubUNITY 142.1.64.254CUPS 142.1.64.254IPCC 142.1.64.254UNITY EXPRESS v9 142.102.64.254 v10 看题2.4 Note1.Check service2.NTP3.HOST -> IP4.企业参数M Group3Cisco U nified Communication Manager3.11CUCM IP Phones registration (3points)Register IP phones at HQ, SiteB and SiteC to CUCM and assign extension numbers as specified in theabove 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 Phone 1. SiteB Phone I etc. IP Phones should display globalizeddialing number at the right hand corner e.gH Q? Phone I should display+14082022001, SiteB Phone Ishould display +19723033001SiteC Phone I should display +852********.Phones should display 12hours time format. Date displayed should be in the format mm-dd-yy.3.2Change background (2points)HQ Phone 1/2/3 SiteB Phone 1/2Can change the default background to that on the lab PC7975 : Desktops/320x216x16/7965:Desktops/320x212x167971 :Desktops/320x212x127961& 7942: Desktops/320x196x4CIPC: Desktops/320x212x12<CiscoIPPhoneImageList><ImageItem Image="TFTP:Desktops/320x216x16/small.png"URL="TFTP:Desktops/320x216x16/large.png"/></CiscoIPPhoneImageList>3.3CUCME IP Phones registration (3points)4001 4002 register to SiteC routerProvide a share line instance with DN number of 4003 on the second line of each of the SiteC Phone.3.4CUCME IP Phones Feature (3points)configuration the CUCME such that the two SiteC phones can accommodate up to 5 inbound to this share number,howeverSiteC phone 1 can only answer a more than of 2 inbound call.On its shard line and SiteC phone 2 can answer up to 4 on its share line.SCphone 1 & 2recieve more than two callsAll allow with phone to display calling information (calling number and name ) for active call on the share linehowever configuration a private button on the third line SiteC phone 1Which the phone user configuration, Press to enable or disable other this share line, phones to display thecoun?? Information for active call SiteC phone's share line.ephone-dn 1octo-linenumber 4001 no-reg primarydescription +852********name SC Phone 1ephone-dn 2octo-linenumber 4002 no-reg primary name SC Phone 2!!ephone-dn 3octo-line number 4003 no-reg primary huntstop channel 5!!ephone 1privacy offprivacy-buttonmac-address 001E.378A.F741 busy-trigger-per-button 2 type CIPCbutton 1:1 2:3ephone 2privacy offmac-address 0017.E0AE.5001 busy-trigger-per-button 4 type 7961button 1:2 2:3!4Voice Gateways and Signaling4.1HQ lOS MGCP TI-PRI gateway (2 points)Configure CUCM to register HQ Router controller TI 0/0/0 as lOS 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 numbers.Calls made to this number should display 10-digit caller ID as 408202xxxx. There is no need to test 9911Calling.ccm-managerconfig server 142.102.64.12 142.102.64.11 ccm-managerconfignoccm-manager configR1(config)#voice-port 0/0/0:23R1(config-voiceport)#shutdownR1(config-voiceport)#int s0/0/0:23R1(config-if)#shR1(config-if)#no isdn bind-l3 ccm-managerR1(config)#controller t1 0/0/0R1(config-controller)#shR1(config-controller)#no pri-group timeslots 1-23R1(config-controller)#pri-group timeslots 1-3 service mgcpR1(config-controller)#R1(config-controller)#no shR1(config-controller)#int s0/0/0:23R1(config-if)#isdn bind-l3 ccm-managerR1(config-if)#isdn outgoing display-ieR1(config-if)#isdn send-alertingR1(config-if)#isdn bchan-number-order ascendingR1(config)#voice-port 0/0/0:23R1(config-voiceport)#no shR1(config)#mgcp bind media source-interface f0/0.103R1(config)#mgcp bind control source-interface f0/0.103card type T1 0 34.2SiteBlOS H.323 TI-PRI gateway (2 points)Configure CUCM to register SiteB Router controller TI 0/0/0 as lOSh323TI PRI gateway.Make sure that all inbound and outbound h323 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 numbers.Calls made to this number should display 10-digit caller ID as 972303xxxx. There is no need to test 9911 calling.interface Vlan303ip address 142.102.65.254 255.255.255.0h323-gatewayvoip interfaceh323-gatewayvoip bind srcaddr 142.102.65.254!4.3SiteClOS H323 gateway (2 points)Configure SiteC router as H323 gateway and register the same to CUCM. Use only 12 channels of El 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 extension999 no need 99994.4 Gatekeeper Request (3 points)Configuration R1 as a GK, domain name Do not register any E.164 number.R1#show gatekeeper endpointsGATEKEEPER ENDPOINT REGISTRATIONCallSignalAddr Port RASSignalAddr Port Zone Name Type Flags 142.100.64.11 1720 142.100.64.1132814 GK VOIP-GWH323-ID: HQ-Trunk IVoice Capacity Max.= Avail.= Current.= 0142.100.64.12 1720 142.100.64.12 32785 GK VOIP-GWH323-ID: HQ-Trunk 2Voice Capacity Max.= Avail.= Current.= 0GATEKEEPER ENDPOINT REGISTRATIONCallSignalAddr Port RASSignalAddr Port Zone Name Type Flags 142.100.64.11 1720 142.100.64.1132814 GK VOIP-GWH323-ID: HQ-Trunk IVoice Capacity Max.= Avail.= Current.= 0142.100.64.12 1720 142.100.64.12 32785 GK VOIP-GW H323-ID: HQ-Trunk 2Voice Capacity Max.= Avail.= Current.= 0142.102.66.254 1720 142.102.66.254 53256GK h323-GW H323-ID: CME-HKGVoice Capacity Max.= Avail.= Current.= 0Total number of active registrations = 3R1#show gatekeeper gw-type-prefixGATEWAY TYPE PREFIX TABLEPrefix: 852*Zone GK master gateway list:142.102.66.254:1720 CME-HKG Prefix: 1*Zone GK master gateway list:142.100.64.11:1720 HQ-Trunk 1142.100.64.12:1720 HQ Trunk 2Section 4: CUCM Call RoutingPSTN access code for all IP phones- 9Country code for US — 1142.102.66.254:1720 CME-HKG Prefix: 1*Zone GK master gateway list:142.100.64.11:1720 HQ-Trunk 1142.100.64.12:1720 HQ Trunk 2Section 4: CUCM Call RoutingPSTN access code for all IP phones- 9Country code for US — 1Country code for Hong Kong - 852National code for HQ and SiteB IP phones — 1International code for HQ and SiteB IP Phones — 011 International code for SiteC IP Phones — 00Note:5Call Routing5.1CUCM Call Routing — HQ Gateway (3 points)HQ PSTN provider specifications are as follows,I) 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 calls, National for long distance calls, 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“852********” 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,All HQ IP phones can make local PSTN calls by dialing 9 followed by 7 digit PSTN number.Calls 911 by dialing 10 digit.1)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 digits 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 by011 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) 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.10-digit Calling number 408202xxxx should be sent out to PSTN along with calling name.4) Configure local route group for both the type of calls mentioned above so that it uses onlyHQ gateway for call routing.5.2CUCM Call Routing — SiteB Gateway (3 points)siteb calling + HQ Site B PSTN providerspecifications are as follows,1) Site B PSTN provider uses leading digits in the called number to signalnon-local calls. 1 for national and 011 for international calls.2) “Called party number” and “called party number type” information must be set in ISDN setupmessages.(Subscriber for local calls, National for long distance calls, and International for International calls).3) If SiteB Phone I makes international call to SiteC Phone 1, 901185224044001service 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. Calls 911 by dialing 10 digit.2) Local and international calls should use SiteB gateway first and then HQ gateway if local SB gatewayis unavailable.When making the local call uses the HQ GW, the calling number should be 10-digits.International call should send calling number +1972303xxxx5.3CMECall Routing - SiteC Gateway (4 points)SiteC PSTN provider specifications are as follows1) 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 setupmessages.(Subscriber for local, National for long distance and International for International calls).3) If SiteC Phone 1 makes international call to HQ Phone 190014082022001 ,service provider expects “14082022001” in called partynumber 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 999 emergency calls. Calling display 8 digit number.By considering the above specifications, configure following requirements,1) All SiteC IP phones can make local PSTN calls by dialing 9 followed by 8-digit PSTN number. For such local calls. PSTN should send 8-diciit callinq number2404xxxx along with calling name. Also, “called party number type” should be set to subscriber for thesecalls. 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 andvariable length dialing digits. Calling number for such calls should be Hong Kong country code and.Example: 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.3) Configure local route group for both the type of calls mentioned above so that gateway for call routing.4) International calls from SiteC IP phones should be routed via SiteC gateway first as local calls.If SiteC gateway is not available, it should use SiteB gateway. (Example 442085554444 -international call need backup)5.4Gatekeeper Routing (4 points)Do not use any default technology prefix aliase or banil statement to muniment the gatekeepr’s channelselectionReference out from Gakekeeper 4.3 for zone prefix and technology prefix informationUse 4-digit dial between the HQ site and SiteC users dial four digit with altering digit “4”, reversedirection users also dial 4 digit string with the first digit begin “2 for now you are not required toconfiguration 4-digit call between SiteB and SiteC.if a 4 digit call (Between the HQ and SiteC) is reject by gatekeeper for any reason it should still routeas international call via pstn.5.5Gatekeeper Routing Troubleshooting (5 points)The HQ user makes frequent internal call to very import U.K.Customers want to reduce the cost of these international calls.The company has signed a contract with a VoIP service provide to route all U.K.Boundary international calls via the provider's h323 network using RAS The CUCM must send all HQ original boundary intermational (Country Code 44) to the local Gatekeeper (Configuration on RI)RI will then forward these calls to the service provider's gatekeeper. Which match the North America internationalaclts?? Code plus the U.K country code (01144) and router the call.Here are same informational regard the service provider's Gatekeeper: Gatekeeper lD:BBGKGatekeeper Domain Name:Gatekeeper IP:157.26.1 .253Your colleague had configuration the solution but could not get it to work. The removed theseconfiguration escalated to you Reconfigure the Gatekeeper Routing on CUCM and Ri .Gath andanalyzer debugs and troubleshooting this issue. Lastly write a short(up to 50 word) to summarize.You finding use “Notepad” on your pc to document your finding as to why is call failing.5.6CUCM Call Routing —“+“ dialing consideration (3 points) Configure CUCM to deliver globalised dialing pattern for SiteB IP ohones. 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 SiteB IP Phone 2 3033002 from SiteB PSTN phone 5252222.2) On SiteB IP phone 2, it displays 7 digit calling number 5252222 along with calling name as“SiteB PSTN”. Do not answer this call.3) Press directories button to go to missed call menu. After selecting missed calls menuThis call should display globalized calling number +19725252222.4) Select this call from list and click dial button to call this number. This should select SiteB gateway for call routing.5) Once call is connected, it should show “To 5252222” on SiteB IP phone 2 display and “From 3033002” on PSTNphone display.6) If SiteB gateway is not available, call should be routed using HQ gateway.10-digit Calling number 972303xxxx should be sent out to PSTN along without calling name.6Codec Selection (2 points)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 useG729 codec.7Media Resource Management7.1MOH (3 points)When SiteB and SiteC IP phones or PSTN users are put on hold, configure local routers to stream G71 1 multicastMOH from router flash. You can use “music-on-hold.au” file in router flash for this multicast requirement.7.2Call Park (3 points)Configure Call Park 2900-2902Make sure Call Park works on both CUCM's for failover.7.3Cisco Unified CallManager Express cBrage (2 points)Enable cBrage on CUCME,so that sitec phone 1 and phone 2 can barge into an active call on the share line.8QoSIt 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.8.1Switch QoS (3 points)1> Map "COS 5" to DSCP value of EF2> On port fa1/0/13 which is connected to HQ Phone 1, guarantee 32k for incomingSCCP signlingtraffic.Excess traffic should be marked to DSCP 8 and thentransmitted By default,ip Phones mark SCCP signaling traffic to CS3.8.2Link fragmentation and interleaving (2 points)There is 384 k frame-relay PVC between HQ and Site B. Configure R1 and R2 to enable MLPP,link fragmentation and interleaving on this circuit.There is 768 k frame-relay PVC between HQ and SiteC. Configure R1 and R3 to enable MLPP,link fragmentation and interleaving on this circuit. fragment delay 10ms interface Virtual-Templatel9Voice Mail IntegrationYou 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.9.1Cisco Unity Connection Integration and Configuration (6 points)Cisco Unity Connection is pre-configured and integrated with CUCM with following configuration,Voicemail Pilot - 2220Voicemail ports - 2221-2224MWlon 1998MWI off- 1999AXL username — administratorAXL password - ccievoiceAdd AXL server then import hql and siteb1/2 to UnityCn (Username: HQ1 SiteB1 I SiteB2)Set default PIN to “246810” for these users.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 seconds9.2Cisco Unity Express Initial Configuration (2 points)9.3Cisco Unity Express and CUCME (2 points)Before proceeding to the cue-cucme integration first check to see if the install CUE license file is for CUCM or CUCME,If the license file is for CUCM you will need to update it to support CUCME integration.FTP server:142.100.64.20Username/pass cisco/cisco(Note Differente for capitalization)The following file are available:CU Exxxxxx.liceCU Exxxxxx.liceuse the following information for cue and cucme9.4Live Recording (5 points)Provision a Live record solution for the SiteCip phone so that initiators of conference calls could invoke a record session to record conversational as voice message into cue.use directory number 4250 as pilot number.10UCCX10.1I CD (5 points)UCCX is pre-configured and integrated with CUCM with below details:lCD Route Point:2400CTI Ports:2401 -2405JtapiUsername:jtapiJtapiPassword:ciscoRmCmUsername:rmRmCmPassword:ciscoUCCX Application UserName:uccxadminUCCX Application Password:ccievoiceUCCX Server UserName:administratorUCCX Server password:ccievoice10.2A dvance要求先听见一个队列宣告,然后宣告你前面还有X位.All other calls into the queue can be configure the cucm,uccx and passed call serviced by either again11: Cisco Unified Presence11.1C UCM presence using busy lamp field (BLF) (3 points)10.2 CUCME Presence (3 points)User the 3th button of SiteC Phone 2 to monitor line 1 user of phone 1.This button on SiteC phone2 should be lit solid Red LED when SiteC Phone 1 is off-hook or Do Not Direct (DND) mode.12: High AvailabilityON SITEB.1. SRST call routing - 2 pointsa. calls should work as normal and should display the following ani when calling the PSTN. The PSTN should also be able to call inbound911 =====> 10 digit displaylocal =====> 7 digit displayld =======> 10 digitinternational ======> +11 digit displayb.When in SRST ,SB Phone can dial HQ Phones using 4 digits ,assuming the telphone service provide support 4 digit.2. Advance SRST - 4 pointsa. 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 phonesb. both of the primary lines on each phone sbph1 and sbph2 should be able to make or recieve more than two callsc. when in SRST phone should still be able to utilize Cbarged. your are not allowed to have information for any ephones in the running configuratione.When in SRST, HQ and SiteC can dial SiteB Phone(3XXX)。