CCRG-4-1-2 (1)

VDC • Create a new virtual context called “VDC2” • Interfaces e4/1, 7/1, and 7/3 added to VDC2hostname RHGvdc RHG.VDC2allocate interface ethernet 4/1, ethernet 7/1, ethernet 7/3show vdc RHG.VDC2 membershipshow vdc RHG.VDC2 • To access the virtual context for further network configurationswitchto vdc RHG.VDC2 • Connect back to the main/global contextswitchback Configuration Reference Guide | [C] 131

Jumbo Frame Support for Nexus 5000 Series • Enable Jumbo frame support on Cisco Nexus 5000 seriespolicy-map type network-qos jumboclass type network-qos class-default mtu 9216system qosservice-policy type network-qos jumboJumbo Frame Support for Nexus 7000 Series • Enable Jumbo frame support on Cisco Nexus 7000 seriessystem jumbomtu 9216 Configuration Reference Guide | [C] 132

OTV • Requires the Transport Services license to be installed • Extend VLANs 10 and 11 between the two sites • The Overlay VLAN will be 100 used between the two sites for extending VLAN10-11>> NX-1 <<interface ethernet 1/1ip address 10.1.1.1/30ip igmp version 3no shutdownvlan 10-11vlan 100feature otvotv site-vlan 100interface Overlay1otv control-group 239.1.1.1otv data-group 232.1.1.0/28otv join-interface ethernet 1/1otv extend-vlan 10-11no shutdownshow otvshow otv overlayshow otv adjacencyshow otv siteshow otv vlanshow otv arpshow mac address-table Configuration Reference Guide | [C] 133

QoS for 48-Port GE (1P3Q4T) • Recommended QoS for Nexus 7000 48-Port 10/100/1000 Modules (1P3Q4T) • CoS to queue mapping for all interfaces with same hardware ASIC • BW, queue-limit and drop threshold are performed using the policy-map • Apply to all interfaces with same hardware ASICclass-map type queuing match-any 1p3q4t-out-pq1match cos 5class-map type queuing match-any 1p3q4t-out-q2match cos 3,6-7class-map type queuing match-any 1p3q4t-out-q3match cos 2,4class-map type queuing match-any 1p3q4t-out-q-defaultmatch cos 0-1policy-map type queuing GE-outboundclass type queuing 1p3q4t-out-pq1 priority level 1 queue-limit percent 15class type queuing 1p3q4t-out-q2 queue-limit percent 25 queue-limit cos 6 percent 100 queue-limit cos 7 percent 100 queue-limit cos 3 percent 70 class type queuing 1p3q4t-out-q3 queue-limit percent 25 queue-limit cos 4 percent 100 queue-limit cos 2 percent 50 bandwidth remaining percent 33class type queuing 1p3q4t-out-q-default queue-limit percent 35 queue-limit cos 1 percent 50 queue-limit cos 0 percent 100 bandwidth remaining percent 45Nexus 7K(config-if)#service-policy type queuing output GE-outbound Configuration Reference Guide | [C] 134

Fibre Channel over Ethernet (FCoE) • Configuration for Nexus 5000 Series! Enable FCoE on the Nexus switchfeature fcoe! Map a VSAN for FCoE traffic onto a VLANvlan 10fcoe vsan 110! Create virtual Fibre Channel interfaces to carry the FCoE trafficinterface vfc 1bind interface ethernet 1/10no shutdown! Map VSAN to vFC interfacevsan databasevsan 110 interface vfc 1! Monitoring FCoEshow flogi database Configuration Reference Guide | [C] 135

Fibre Channel • Configuration for Nexus 5548UP series • Enable ports 28 to 32 on Nexus switch as Fibre Channel (fc) portsslot 1 port 28-32 type fc Configuration Reference Guide | [C] 136

Solution/Services: Voice & Unified CommunicationsRelated: Cisco Unity ExpressBase Configuration • Configure VLAN and interfaces for Voice network. Voice VLAN will be VLAN10 (192.168.10.0) • “ip source-address”: Specify source IP address for CME and list SCCP port number (2000) • “max-ephone”: specify the maximum number of phones supported on the router • “max-dn”: specify the maximum number of directory numbers supported on router • “timeout interdigit”: specify the amount of time (in seconds) for setting up a call • “system message”: configure banner on bottom of the phone • “video”: enable video support for phone endpoints • “time-zine”: specify correct time-zone. 5 = PST (GMT -8) • “voicemail”: specify the main voicemail pilot number. • “web admin system”: configures username and password to access the CME Integrated GUI pagevlan 10name ROUTEHUB-VLANinterface FastEthernet0/1/1description IP Phone Portswitchport access vlan 10interface Vlan10ip address 192.168.10.1 255.255.255.0telephony-serviceip source-address 192.168.10.1 port 2000max-ephones 14max-dns 56timeouts interdigit 5 Configuration Reference Guide | [C] 137

system message RouteHub UC520auto assign 19 to 19videotime-zone 5voicemail 6000web admin system name admin secret cisco123moh music-on-hold.aucall-forward pattern .Tcall-forward system redirecting-expandedtransfer-system full-consult dsstransfer-pattern 9.Tsecondary-dialtone 9Directory Numbers • Configure directory number (extension) 6700 • Configure extension as a dual-line number to accept up to 2 concurrent calls • Any call to extension 6700 that is “busy” or there is “no answer (noan)” send to voicemail (6000) • “label”: specify what text will be displayed on the phone for the configured extension.ephone-dn 10 dual-linenumber 6700label 6700 (Main)description 2091236700call-forward busy 6000call-forward noan 6000 timeout 15IP Phone Configuration • Add Cisco IP 7970 Phone using mac-address 001C.58F0.7619 • Associate the extension (6700) in “ephone-dn 10” to line 1 on the Cisco phoneephone 1 mac-address 001C.58F0.7619 type 7970 button 1:10 Configuration Reference Guide | [C] 138

Call Forward All • All calls made to extension 6700 will be forwarded to 4001ephone-dn 10 dual-linenumber 6700call-forward all 4001Voice and Data VLAN Configuration • Voice VLAN = 10, Data VLAN = 100vlan 10name RHG-VOICE-VLANvlan 100name RHG-DATA-VLANinterface FastEthernet0/1description TO: IP Phone and Desktopswitchport access vlan 100switchport mode accessswitchport voice vlan 10spanning-tree portfastinterface GigabitEthernet0/1description TO: UPLINK (Core, Distribution)switchport trunk allowed vlan 10,100switchport mode trunkswitchport nonegotiatespanning-tree portfast trunk Configuration Reference Guide | [C] 139

Configuring DHCP on Cisco IOS • Configure DHCP scope for Voice Network (192.168.10.0/24) • Configure “option 150” to point to the CME router IP (192.168.10.1)ip dhcp excluded-address 192.168.10.1 192.168.10.29ip dhcp pool ROUTEHUB-DHCP-LAN-POOLnetwork 192.168.10.0 255.255.255.0default-router 192.168.10.1option 150 ip 192.168.10.1dns-server 4.2.2.2 4.2.2.3lease 5Mapping an Analog Line (DID) to an IP Phone • When user at extension 6700 dials any outgoing call it will route through FXO port 1/0/0 ; dedicated for 6700 • All incoming calls on FXO port 1/0/0 would go to extension 6700voice translation-rule 1rule 1 /^9/ /19/voice translation-profile TP-6700translate called 1dial-peer voice 1 voiptranslation-profile incoming TP-6700answer-address 6700dial-peer voice 19 potsdestination-pattern 19Tport 1/0/0voice-port 1/0/0supervisory disconnect dualtone pre-connectpre-dial-delay 0no vadtimeouts call-disconnect 2timeouts wait-release 2connection plar opx 6700caller-id enable Configuration Reference Guide | [C] 140

Configuring FXS port as a SCCP port • Configure FXS port 0/0/0 as a SCCP port • Plug analog phone into FXS pot 0/0/0. Type: ANL, MAC Address: determine from the command “show stcapp device summary”sccp local Vlan10sccp ccm 192.168.10.1 identifier 1 priority 1 version 4.1sccpsccp ccm group 1bind interface Vlan10associate ccm 1 priority 1keepalive retries 5switchback method gracefulstcapp ccm-group 1stcappdial-peer voice 14 potsservice stcappport 0/0/0voice-port 0/0/0 caller-id enableephone 2device-security-mode nonemac-address D456.7C69.0000type anlbutton 1:10 Configuration Reference Guide | [C] 141

CME as SIP Server for SIP Clients • Enable SIP • Configure CME as SIP Server (using 192.168.10.1) for SIP IP Phones • Add new directory number (extension) 8700 used for SIP Phones • Associate extension 8700 to SIP phone (using register pool 1) with id “000C.F179.1682” • SIP phone will require using account “8700” for the username and password being “cisco6778”voice service voipallow-connections h323 to sipallow-connections sip to h323allow-connections sip to sipsupplementary-service h450.12voice register globalmode cmesource-address 192.168.10.1 port 5060max-dn 12max-pool 12timezone 47time-format 24date-format YY-M-Ddst start Oct week 8 day Sun time 02:00dst stop Mar week 8 day Sun time 02:00voice register dn 1number 8700name ROUTEHUB SIP client (X-lite)voice register pool 1id mac 000C.F179.1682number 1 dn 1username 8700 password cisco6778codec g711ulaw Configuration Reference Guide | [C] 142

Blocking Incoming Calls from PSTN • Block outside number “1-800-123-4567” from coming into CME router FXO port 0/1/0voice translation-rule 5rule 1 reject /8001234567/voice translation-profile call_blocktranslate calling 5dial-peer voice 100 potscall-block translation-profile incoming call_blockcall-block disconnect-cause incoming call-rejectdestination-pattern 9.Tincoming called-number .port 0/1/0Phone Directory • Phone directory will list entries based on first name • Phone directory entries reflecting the speed-dial number and labeltelephony-servicedirectory first-name-firstdirectory entry 1 919252302203 name ROUTEHUB (Main)directory entry 2 912091234567 name Other Number (Cell)Single Number Reach (SNR) • If some calls extension 6700 it will also ring the number 919252302203 (access code is 9 for outbound calls) • If there is no answer, the call will be forwarded to internal voicemail (6000)ephone-dn 10 dual-linenumber 6700 no-reg primarymobilitysnr 919252302203 delay 2 timeout 30 cfwd-noan 6000 Configuration Reference Guide | [C] 143

Fast Dial • Configure fast dial entry 6702 (USER2) on the phone configured under “ephone 1”ephone 1fastdial 1 6702 name USER2Call Park • Call Park number will be 6002 • A call can be parked for 30 seconds and support up to 10 concurrent parked callsephone-dn 14number 6002park-slot timeout 30 limit 10name ROUTEHUB CALL PARK Configuration Reference Guide | [C] 144

Setting up SIP Trunk to SIP Provider (with CME/CUE) • IOS: 15.1 • “allow-connections sip to sip”: Enable SIP connections to/from CME router “ • Configure SIP trunk to SIP provider (sipproxy.routehub.local) on Internet. • Our SIP phone number will be “19252302204” • Our SIP username will be our SIP number and our SIP password will be “cisco6778”voice-card 0dspfarmdsp services dspfarmsip-uaauthentication username 19252302204 password cisco6778no remote-party-idretry invite 2retry register 10timers connect 100registrar dns:sipproxy.routehub.local expires 3600sip-server dns:sipproxy.routehub.localhost-registrarvoice service voipip address trusted listipv4 0.0.0.0 0.0.0.0allow-connections sip to sipsupplementary-service h450.12no supplementary-service sip moved-temporarilyno supplementary-service sip refersip registrar server expires max 3600 min 3600 localhost dns:sipproxy.routehub.local • Configure translation rule for incoming calls to translate SIP number to extension 6700 which is configured on a phonevoice translation-rule 1rule 1 /19252302204/ /6700/voice translation-profile RHG-TP-SIP-INtranslate called 1 Configuration Reference Guide | [C] 145

• Configuring translation for outgoing calls to (1) allow users to dial 911, (2) any call placed over the SIP trunk will strip off the access code “9” before routing the call, and (3) any extension or DID placing a call through the SIP trunk will be translated to our SIP phone number (show in translation rule #3)voice translation-rule 2rule 1 /^911$/ /911/rule 2 /^9\(.*\)$/ /\1/voice translation-rule 3rule 1 /^.*/ /19252302204/ • Translation rule used for outgoing calls from Cisco Unity Express (CUE): (1) local calls will automatically include the local area code, (2) calls from directory number 6001 (the AA pilot) will map to DID 925-230-2204, (3) calls from directory number 6000 (the Voicemail pilot) will map to DID 925-230-2204, and (4) any call placed over the SIP trunk will strip off the access code “9” before routing the call.voice translation-rule 4rule 1 /^9\(.......\)$/ /925\1/rule 2 /6001/ /19252302204/rule 3 /6000/ /19252302204/rule 4 /^9\(.*\)$/ /\1/voice translation-profile RHG-TP-SIP-OUTtranslate called 2translate calling 3translate redirect-target 4translate redirect-called 4voice translation-profile RHG-TP-SIP-CUEtranslate redirect-target 4translate redirect-called 4 • Specify supported codec to use with the SIP trunkvoice class codec 1codec preference 1 g711ulaw Configuration Reference Guide | [C] 146

• Dial peer for incoming calling from SIP trunkdial-peer voice 1 voipdescription INCOMING: All Incoming Callingtranslation-profile incoming RHG-TP-SIP-INdestination-pattern .%voice-class codec 1voice-class sip dtmf-relay force rtp-ntesession protocol sipv2session target sip-serverdtmf-relay rtp-nteno vad • Dial peer for 7-digit Local calling through SIP trunkdial-peer voice 7 voipdescription OUTGOING: 7-Digit Local Callingtranslation-profile outgoing RHG-TP-SIP-OUTdestination-pattern 9[2-9]......voice-class codec 1voice-class sip dtmf-relay force rtp-ntesession protocol sipv2session target sip-serverdtmf-relay rtp-nteno vad • Dial peer for Long Distance calling through SIP trunkdial-peer voice 11 voipdescription OUTGOING: Long Distance Callingtranslation-profile outgoing RHG-TP-SIP-OUTdestination-pattern 91[2-9]..[2-9]......voice-class codec 1voice-class sip dtmf-relay force rtp-ntesession protocol sipv2session target sip-serverdtmf-relay rtp-nteno vad Configuration Reference Guide | [C] 147

• Dial peer for Emergency 911 calling through SIP trunkdial-peer voice 9911 voipdescription OUTGOING: Emergency Callingtranslation-profile outgoing RHG-TP-SIP-OUTdestination-pattern 9911voice-class codec 1voice-class sip dtmf-relay force rtp-ntesession protocol sipv2session target sip-serverdtmf-relay rtp-nteno vad • Dial peer for Emergency 911 calling through SIP trunkdial-peer voice 911 voipdescription OUTGOING: Emergency Callingtranslation-profile outgoing RHG-TP-SIP-OUTdestination-pattern 911voice-class codec 1voice-class sip dtmf-relay force rtp-ntesession protocol sipv2session target sip-serverdtmf-relay rtp-nteno vad • Dial peer for International calling through SIP trunkdial-peer voice 9011 voipdescription OUTGOING: International Callingtranslation-profile outgoing RHG-TP-SIP-OUTdestination-pattern 9011Tvoice-class codec 1voice-class sip dtmf-relay force rtp-ntesession protocol sipv2session target sip-serverdtmf-relay rtp-nteno vad Configuration Reference Guide | [C] 148

• Dial peer for Voicemail (on CUE)dial-peer voice 6000 voipdescription CUE: Voicemailtranslation-profile outgoing RHG-TS-SIP-CUEdestination-pattern 6000b2buasession protocol sipv2session target ipv4: 192.168.10.2dtmf-relay sip-notifycodec g711ulawno vad • Dial peer for AA (on CUE)dial-peer voice 6001 voipdescription CUE AAtranslation-profile outgoing RHG-TS-SIP-CUEdestination-pattern 6001b2buasession protocol sipv2session target ipv4: 192.168.10.2dtmf-relay sip-notifycodec g711ulawno vad • MWI notificationdial-peer voice 800 voipincoming called-number 800[0,1]....codec g711ulawno vadephone-dn 16number 8000.... no-reg primarymwi onephone-dn 17number 8001.... no-reg primarymwi off Configuration Reference Guide | [C] 149

• Directory number 6700 used for placing and receiving calls from the SIP trunkephone-dn 10 dual-linenumber 6700 no-reg primaryname 6700call-forward busy 6000call-forward noan 6000 timeout 15ephone-dn 18number 9252302204description “Main Number” • Preserves caller-ID of a call when it is transferred or forwarded • Enable translation rule features for forwardingtelephony-servicecalling-number initiatorcall-forward system redirecting-expanded • CUE configuration on CME routerinterface Integrated-Service-Engine0/0description RHG: CUE interfaceip unnumbered Vlan10service-module ip address 192.168.10.2 255.255.255.0service-module ip default-gateway 192.168.10.1ip route 192.168.10.2 255.255.255.255 Integrated-Service-Engine0/0 • Monitoring commandsshow sip-ua register statusdebug ccsip messageshow ephone registeredshow voice rtp connectionshow sip-ua callshow call active voice briefdebug ccsip message Configuration Reference Guide | [C] 150