Lab 4-Solutions Guide

2. Basic Campus Design 2.1 Voice and Data VLANs

Configure Voice VLANs for switch ports connecting to IP Phones at HQ, SiteB and SiteC. 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 placed in 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 Solution: HQ 3750 Switch HQ-SW(confg)# cdp run HQ-SW(confg)# cdp advertise-v2 Verify the vlan configuration in 3750 switch HQ-SW# sh vlan br VLAN Name Status Ports ---- -------------------------------- --------- ------------------------------1 default active Gi0/2, Gi0/3, Gi0/4, Gi0/13 100 Servers active Gi0/1 102 HQ-Voice active Gi0/5, Gi0/6, Gi0/7, Gi0/8 202 HQ-Data active Gi0/5, Gi0/6, Gi0/7, Gi0/8 1004 fddinet-default act/unsup 1005 trbrf-default act/unsup If vlan is not there create the required vlan as follows HQ-SW(config)#vlan 102 HQ-SW(config-vlan)#name HQ-Voice Configure the interface connected to HQ Router interface FastEthernet1/0/2 description ** HQ-Router ** switchport trunk native vlan 202 switchport trunk encapsulation dot1q switchport trunk allowed vlan 100,102,202 switchport mode trunk

Configure the interface connected to phone interface FastEthernet1/0/5 description HQ Phones switchport access vlan 102 switchport mode access switchport voice vlan 202 spanning-tree portfast Configure the interface connected to server interface FastEthernet0/1 description ** Server ** switchport mode access switchport access vlan 100 mls qos trust dscp spanning-tree portfast ip dhcp snooping vlan 102 no ip dhcp snooping information option /* if dhcp snooping is enabled */ ip dhcp snooping ip arp inspection vlan 102 HQ-RTR config interface GigabitEthernet0/1.100 encapsulation dot1Q 100 ip address 142.100.64.254 255.255.255.0 ! interface GigabitEthernet0/1.102 encapsulation dot1Q 102 ip address 142.102.64.254 255.255.255.0 ip helper-address 142.100.64.11 ! interface GigabitEthernet0/1.202 encapsulation dot1Q 202 native ip address 142.202.64.254 255.255.255.0 SiteB Router (4ESW) module Verify the vlan configuration in SiteB router. If there is no vlan available create the vlan R2#sh vlan-sw br VLAN Name Status Ports ---- -------------------------------- --------- ------------------------------1 default active Fa1/1, Fa1/2, Fa1/3, Fa1/4 302 Voice active 402 Data active 1005 trbrf-default act/unsup

R2#vlan database R2(vlan)#vlan 302 name SiteB-Voice VLAN 302 modified: Name: SiteB-Voice R2(vlan)#exit APPLY completed. Exiting.... Verify the layer 3 interfaces are up R2#sh ip int br Interface IP-Address OK? Method Status Serial0/0/1:0 unassigned YES NVRAM up Serial0/0/1:0.101 142.1.67.2 YES NVRAM up Vlan1 unassigned YES NVRAM up Vlan302 142.102.65.254 YES NVRAM up Vlan402 142.202.65.254 YES NVRAM up SSLVPN-VIF0 unassigned NO unset up Loopback0 142.1.65.254 YES NVRAM up

Protocol up up up up up up up

If layer 3 vlan interface is showing protocol down make sure the vlan is assigned to the phone connected interface. If not, Configure the phone connected interface as below interface FastEthernet1/0 switchport trunk native vlan 402 switchport mode trunk switchport voice vlan 302 SiteC Router (4ESW) module Verify the vlan configuration in SiteC router. If vlan is not there create the required vlan. R3#sh vlan-sw br VLAN Name Status Ports ---- -------------------------------- --------- ------------------------------1 default active Fa1/1, Fa1/2, Fa1/3, Fa1/4 502 Voice active 602 Data active R2#sh cdp Global CDP information: Sending CDP packets every 60 seconds Sending a holdtime value of 180 seconds Sending CDPv2 advertisements is enabled

R3#vlan database R3(vlan)#vlan 502 name SiteC-Voice VLAN 502 modified: Name: SiteC-Voice R3(vlan)#exit Verify the layer 3 interfaces are up R3#sh ip int br Interface IP-Address OK? Method Status Serial0/0/1:0 unassigned YES NVRAM up Serial0/0/1:0.101 142.1.67.2 YES NVRAM up Vlan1 unassigned YES NVRAM up Vlan502 142.102.66.254 YES NVRAM up Vlan602 142.202.66.254 YES NVRAM up SSLVPN-VIF0 unassigned NO unset up Loopback0 142.1.66.254 YES NVRAM up R3#

Protocol up up up up up up up

If layer 3 vlan interface is showing down make sure the vlan is assigned to the phone connected interface. If not, configure the phone connected interface as below interface FastEthernet1/0 switchport trunk native vlan 602 switchport mode trunk switchport voice vlan 502

2.2

DHCP Service

Configure CUCM Publisher as DHCP server to provide IP Addresses for IP Phones at HQ and SiteB from their respective Voice subnets. For HQ, use IP address range from 142.102.64.10/24 to 142.102.64.30/24 For SiteB, use IP address range from 142.102.65.10/24 to 142.102.65.30/24 Configure local Cisco 2811 router as DHCP server to provide IP addresses for SiteC IP Phones from local Voice subnet. Use IP address range from 142.102.66.10/24 to 142.102.66.30/24

Solution: Configure CUCM Publisher as dhcp server

Configure dhcp subnet for HQ and Site B voice vlan individually HQ voice vlan

SiteB voice vlan

Configure ip helper-address in layer3 interface of the voice vlan in HQ and SiteB router R1# interface GigabitEthernet0/0.102 encapsulation dot1Q 102 ip address 142.102.64.254 255.255.255.0 ip helper-address 142.100.64.11 ! R2# interface Vlan302 ip address 142.102.65.254 255.255.255.0 ip helper-address 142.100.64.11 ! If the phone connected to 3750 switch is not getting the ip address from dhcp server verify DHCP snooping and ARP inspection is configured. If it is, configure the interface connected to router as trusted interface. HQ-SW# interface GigabitEthernet0/1 description ** HQ Router ** ip arp inspection trust ip dhcp snooping trust ! Configure DHCP service in SiteC Cisco 2811 router for SiteC voice vlan Configure the excluded ip address range before configuring the dhcp pool R3# ip dhcp excluded-address 142.102.66.1 142.102.66.9 ip dhcp excluded-address 142.102.66.31 142.102.66.254 ! ip dhcp pool IPT network 142.102.66.0 /24 default-router 142.102.66.254 option 150 ip 142.1.66.254 ! Verification of DHCP Service sh ip dhcp bindings sh arp Debug commands debug ip dhcp server events debug ip dhcp server packet

2.3

NTP

Synchronize HQ router with external NTP source at 157.26.1.100. 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 they belong Solution: Configure the HQ router to synchronize the time with PSTN router IP 157.26.1.100 and configure the branch site routers to synchronize with HQ router. HQ router R1(config)#clock timezone PST -8 R1(config)#clock summer-time PST recurring R1(config)#ntp server 157.26.1.100 Site B router R2(config)#clock timezone CST -6 R2(config)#clock summer-time CST recurring R2(config)#ntp server 142.1.64.254 Site C router R3(config)#clock timezone HKT 8 R3(config)#ntp server 157.26.1.100 R3(config)#clock summer-time HKT recurring Verification R1#sh ntp status Clock is synchronized, stratum 5, reference is 157.26.1.100 nominal freq is 250.0000 Hz, actual freq is 249.9993 Hz, precision is 2**24

CUCM Publisher Configure the CUCM Publisher to synchronize with the external time source and the Subscriber servers in the cluster will synchronize with CUCM Publisher by default. 1. Go to Cisco Unified Operating System Administration -> Settings -> NTP Servers

2. Verify the NTP from SSH CLI & If required restart the NTP service from SSH CLI on Publisher and Subscriber admin:utils ntp status ntpd (pid 22074) is running... remote refid st t when poll reach delay offset jitter

==================================================== 127.127.1.0 *142.1.64.254 LOCAL(0) 10 l 49 5u 64 47 17 64 0.000 17 0.000 0.002

157.26.1.100

0.672 -14.194 15.803

synchronised to NTP server (142.1.64.254) at stratum 6 time correct to within 1019 ms polling server every 64 s Current time in UTC is : Thu Oct 21 15:06:39 UTC 2010 Current time in America/New_York is : Thu Oct 21 11:06:39 EDT 2010 admin: admin:utils ntp restart

3. Cisco Unified Communications Manager 3.1 CUCM & CUCME IP Phone Registrations
Register IP phones at HQ, SiteB to CUCM and assign extension numbers as specified in the above table. Regsiter IP phones at SiteC to CME (local router) 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.g-HQ Phone 1 should display +14082022001, SiteC Phone 1 should display +8522404400 IP Phones should display globalized dialing number at the right hand corner e.g- SiteC Phone 1 should display +85224044001. Solution: Verify the necessary services are started in Publisher and Subscriber Publisher

Subscriber

Configure the Basic Setup of CUCM Verify the servers are configure with ip address instead of host name

Unified CM

Configure Unified CM Group

Configure the Date and Time Group for HQ, SiteB and SiteC HQ

SiteB

List of Date/Time group for all sites

Create Device Pool for all sites with correct Date and Time Group selected

Verify the following Enterpriser Service Parameters

Create partition and which will be assigned to phone DNs and create internal CSS which will be assigned to Gateways in later sections.

Enable Auto Registration of Phone in Primary CPE (Publisher/Subscriber) as mentioned in Unified CM Group Or Manually add the Phones.

Edit the Phone settings and DN settings of all phones as given in the table Example given for hq ph 1 Phone Settings

DN Settings

CUCME Phone Registration R3# ! tftp-server flash:7941-7961/apps41.8-4-1-23.sbn alias apps41.8-4-1-23.sbn tftp-server flash:7941-7961/cnu41.8-4-1-23.sbn alias cnu41.8-4-1-23.sbn tftp-server flash:7941-7961/dsp41.8-4-1-23.sbn alias dsp41.8-4-1-23.sbn tftp-server flash:7941-7961/term41.default.loads alias term41.default.loads tftp-server flash:7941-7961/term61.default.loads alias term61.default.loads tftp-server flash:7941-7961/cvm41sccp.8-4-1-23.sbn alias cvm41sccp.8-4-1-23.sbn tftp-server flash:7941-7961/jar41sccp.8-4-1-23.sbn alias jar41sccp.8-4-1-23.sbn tftp-server flash:7941-7961/SCCP41.8-4-2S.loads alias SCCP41.8-4-2S.loads ! telephony-service no auto-reg-ephone load 7961 SCCP41.8-4-2S max-ephones 2 max-dn 2 no-reg both time-zone 42 time-format 24 date-format mm-dd-yy ip source-address 142.1.66.254 port 2000 max-conferences 12 gain -6 transfer-system full-consult create cnf-files version-stamp 7960 Nov 06 2010 15:31:20 ! ephone-dn 1 dual-line number 4001 description +85224044001 name SiteC Ph1 ! ephone-dn 2 dual-line number 4002 description +85224044002 name SiteC Ph2 ! ephone 1 device-security-mode none mac-address 001B.D4A0.9428 type 7961 button 1:1 ! ephone 2 device-security-mode none mac-address 001B.D4C6.D8B6 type 7961 button 1:2 !

3.2 Cisco CUCM IP Phone Customization

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 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. Solution:

HQ Ph1 and HQ PH2

3.3 CUCME IP Phone Customization

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. Solution: ephone-dn 3 number A5001 intercom A5002 label "SC PH2" ! ephone-dn 4 number A5002 intercom A5001 barge-in label "SC PH1" ! ephone 1 device-security-mode none mac-address 001B.D4A0.9428 type 7961 button 1:1 2:3 ! ephone 2 device-security-mode none mac-address 001B.D4C6.D8B6 type 7961 button 1:2 2:4 !

CUCME IP Phone Customization

Background image on SCPH1 & SCPH2 Images are found on the candidate test pc in the candidate folder on the desktop. Images are call voice-large and voice-small. The proctor said he preferred it was saved on the phone. 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. Solution: To change the background in phones create List.xml file with thumbnail and the real image. Copy the xml file and image file into the same folder on TFTP server and restart the TFTP service on both (PUB & SUB) servers.

Reference: Customizing the Cisco Unified IP Phone chapter in Cisco Unified IP Phone 7965G and 7945G Administration Guide from Maintain and Operate Guides of Cisco web site.
List.xml Example (Image is pointing to Thumb Nail image and URL is pointing to Real image)

<CiscoIPPhoneImageList> <ImageItem Image="TFTP:Desktops/320x212x16/TN-Fountain.png" URL="TFTP:Desktops/320x212x16/Fountain.png"/> </CiscoIPPhoneImageList>

Upload the file to TFTP server from Cisco Unified Operating Systems Administration page. Do the following in both (PUB & SUB) servers.

Verify the files are properly uploaded to TFTP servers

R3#sh flash: /* verify is there image files are available */ R3#del * /* if you want to delete the existing files from directory */ Delete filename [/Desktops/320x196x4/*]? Delete flash:/Desktops/320x196x4/ipxlogo-tn.png? [confirm] Delete flash:/Desktops/320x196x4/ipxlogo.png? [confirm] Delete flash:/Desktops/320x196x4/List.xml? [confirm] Create the following directories R3#mkdir flash:/Desktops/ R3#mkdir flash:/Desktops/320x196x4/ R3#cd flash:/Desktops/320x196x4/ Copy the files from cucm to SiteC Router R3#copy tftp:Desktops/320x196x4/List.xml flash:/Desktops/320x196x4/ Address or name of remote host [142.100.64.11]? Source filename [Desktops/320x196x4/List.xml]? Destination filename [/Desktops/320x196x4/List.xml]? Accessing tftp://142.100.64.11/Desktops/320x196x4/List.xml... Loading Desktops/320x196x4/List.xml from 142.100.64.11 (via Serial0/0/1:0.102): ! [OK - 155 bytes] R3#copy tftp:Desktops/320x196x4/ipxlogo-tn.png flash:/Desktops/320x196x4/ R3#copy tftp:Desktops/320x196x4/ipxlogo.png flash:/Desktops/320x196x4/

Verify the files are there after copy R3#dir flash:/Desktops/320x196x4/ Directory of flash:/Desktops/320x196x4/ 61 -rw2696 Mar 27 2010 07:12:36 +08:00 ipxlogo-tn.png 62 -rw6876 Mar 27 2010 07:13:14 +08:00 ipxlogo.png 63 -rw156 Mar 27 2010 08:01:18 +08:00 List.xml R3(config)#tftp-server flash:Desktops/320x196x4/List.xml R3(config)#tftp-server flash:Desktops/320x196x4/ipxlogo.png R3(config)#tftp-server flash:Desktops/320x196x4/ipxlogo-tn.png

4. Voice Gateways and Signaling 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. Solution: HQ-RTR MGCP Gateway configuration R1(config)#no ip domain lookup R1(config)#ip domain name ccievoice.com R1(config)#network-clock-participate wic 0 R1(config)#network-clock-select 1 t1 0/0/0 R1(config)#isdn switch-type primary-ni controller T1 0/0/0 framing esf linecode b8zs pri-group timeslots 1-24 service mgcp ! interface Serial0/0/0:23 isdn bind-l3 ccm-manager ! ccm-manager switchback immediate ccm-manager redundant-host 142.100.64.11 ccm-manager mgcp

! mgcp mgcp mgcp mgcp mgcp

call-agent 142.100.64.12 service-type mgcp version 0.1 dtmf-relay voip codec all mode out-of-band bind control source-interface GigabitEthernet0/1.102 bind media source-interface GigabitEthernet0/1.102

Add the MGCP gateway into CUCM

Verify the gateway configuration R1#sh isdn status Global ISDN Switchtype = primary-ni %Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 0. Layer 3 output may not apply ISDN Serial0/0/0:23 interface dsl 0, interface ISDN Switchtype = primary-ni L2 Protocol = Q.921 0x0000 L3 Protocol(s) = CCM MANAGER 0x0003 Layer 1 Status: ACTIVE

Layer 2 Status: TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED R1#sh ccm MGCP Domain Name: R1.ccievoice.com Priority Status Host ========================================================= === Primary Registered 142.100.64.12 First Backup Backup Ready 142.100.64.11 Second Backup None R1#

If not showing the correct ip address of the router interface configured for mgcp in CUCM as shown above, do the following: R1(config)#no mgcp R1(config)#no mgcp bind control source-interface GigabitEthernet0/0.102 R1(config)#no mgcp bind media source-interface GigabitEthernet0/0.102 R1(config)#mgcp bind control source-interface GigabitEthernet0/0.102 R1(config)#mgcp bind media source-interface GigabitEthernet0/0.102 R1(config)# R1(config)#mgcp R1(config)# Verify the configuration again and confirm everything is ok as per requirement. Create Separate Route Group for each site to route the Emergency 911 calls. Read the call routing section carefully to understand the requirement of partitions and CSS. Create CSS for HQ, SiteB and SiteC phones such as css-hq-all, css-siteb-all, css-sitec-all and assign them to the respective phones.

For Emergency Calls 1. 2. 3. 4. Create Create Create Create Route Group with HQ Gateway RL with RG-HQ required Partition (pt-hq-911) and assign to the CSS Route Pattern for 911 calls

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 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. Solution: no ip domain lookup ip domain name ccievoice.com ! network-clock-participate wic 0 network-clock-select 1 t1 0/0/0

isdn switch-type primary-ni ! controller T1 0/0/0 framing esf linecode b8zs pri-group timeslots 1-24 ! interface Serial0/0/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice voice isdn outgoing display-ie isdn outgoing ie redirecting-number no cdp enable ! voice class codec 1 codec preference 1 g711ulaw bytes 160 codec preference 2 g729r8 bytes 20 ! voice class h323 1 h225 timeout tcp establish 2 ! voice translation-rule 4 rule 1 /.*\(3$\)/ /\1/ ! voice translation-rule 10 rule 1 /^3...$/ /972303\0/ ! voice translation-profile 911 translate calling 10 ! voice translation-profile did translate called 4 ! voice-port 0/0/0:23 translation-profile incoming did ! dial-peer voice 1 pots incoming called-number . direct-inward-dial port 0/0/0:23 ! dial-peer voice 911 pots translation-profile outgoing 911 destination-pattern 911 no digit-strip port 0/0/0:23 ! dial-peer voice 3002 voip destination-pattern 3...$ voice-class codec 1 voice-class h323 1 session target ipv4:142.100.64.12 dtmf-relay h245-alphanumeric

! dial-peer voice 3001 voip preference 1 incoming called-number . destination-pattern 3...$ voice-class codec 1 voice-class h323 1 session target ipv4:142.100.64.11 dtmf-relay h245-alphanumeric ! interface Vlan302 ip address 142.102.65.254 255.255.255.0 ip helper-address 142.100.64.11 h323-gateway voip interface h323-gateway voip bind srcaddr 142.102.65.254 Note: Add this as H323 gateway into cucm, create RG, RL and RP for emergency calls

4.3 SiteC CUCME E1-PRI gateway

Configure SiteC router as H323 gateway and register the same to CUCM. Use only 12 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. Solution: network-clock-participate wic 0 network-clock-select 1 e1 0/0/0 ! isdn switch-type primary-net5 ! controller E1 0/0/0 pri-group timeslots 1-12,16 ! interface Serial0/0/0:15 no ip address encapsulation hdlc isdn switch-type primary-net5 isdn incoming-voice voice isdn outgoing display-ie isdn outgoing ie redirecting-number ! voice translation-rule 4 rule 1 /.*\(4$\)/ /\1/ ! voice translation-rule 8 rule 1 /^4...$/ /2404\0/ ! voice translation-profile 999 translate calling 8 ! voice translation-profile did translate called 4 ! voice-port 0/0/0:15 translation-profile incoming did ! dial-peer voice 1 pots incoming called-number . direct-inward-dial port 0/0/0:15 ! dial-peer voice 999 pots translation-profile outgoing 999 destination-pattern 999 port 0/0/0:15 forward-digits all

5. CUCM Call Routing 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, PSTN should send 7digit 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. Solution: Create necessary partition and assign to the existing CSS. Create the required Route Patterns with proper called party number type and called party number.

Route Group is created in earlier section so no need to create it again. Create Route List with Standard Local Route Group assigned to RL. Associate the Local Route Group with Device Pool.

Now we can add the Route Pattern for local calls.

Only 7 digit number and name should be displayed to the PSTN phone.

Called party number and called party number type should set here.

Now create the Route Pattern for International calls. We should not use leading digit (011) for international calls, so we must strip the code before sending to PSTN.

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 callsfor 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. Solution: voice translation-rule 7 rule 1 /^3...$/ /303\0/ ! voice translation-rule 9 rule 1 // // type any subscriber plan any isdn ! voice translation-rule 10 rule 1 /^3...$/ /972303\0/ ! voice translation-rule 91 rule 1 // // type any national plan any isdn ! voice translation-profile local translate calling 7 translate called 9 ! voice translation-profile national translate calling 10 translate called 91 ! dial-peer voice 9 pots translation-profile outgoing local destination-pattern 9....... port 0/0/0:23 forward-digits 7 ! dial-peer voice 91 pots translation-profile outgoing national destination-pattern 91408....... port 0/0/0:23 forward-digits 11 !

Create necessary partition and assign to the existing CSS. Create the required Route Patterns with proper called party number type and called party number.

Look at the difference between the HQ and SiteB route pattern for local calls. SiteB route pattern having the second digit of dialed numbers range 0-9 not like HQ which is 2-9. There is no specific numbers mentioned in the place of second digit of route pattern.

We have to work for long distance calls begin with area code 408. The HQ gateway should be act as a primary gateway and SiteB gw should act as a backup gw. So create the separate route list with HQ gw selected as the primary and Siteb gw selected as the backup gw. So do the called party number manipulations in route list details level not in the route pattern like local calls. When the call flow through HQ gw it is going out to pstn as a local call so we have to set the called party number type as subscriber.

When the call flow through SiteB gw it is going out as a long distance call so we have to set the called party number type as national.

GW order listed here having the priority (role of primary and backup gateway) as listed.

Create the route pattern. The call should go through HQ first and SiteB second so we have to strip the national and area code.

5.3 CUCM Call Routing SiteC H323 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 8-digit 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) Solution: voice translation-rule 8 rule 1 /^4...$/ /2404\0/ ! voice translation-rule 9 rule 1 // // type any subscriber plan any isdn ! voice translation-rule 852 rule 1 /^4...$/ /+8522404\0/ ! voice translation-rule 900 rule 1 // // type any international plan any isdn ! voice translation-profile international translate calling 852 translate called 900 ! voice translation-profile local translate calling 8 translate called 9

! dial-peer voice 9 pots translation-profile outgoing local destination-pattern 9[2-8]....... port 0/0/0:15 forward-digits 8 ! dial-peer voice 900 pots translation-profile outgoing international destination-pattern 900T port 0/0/0:15 !

5.4 CUCM Call Routing + dialing consideration

Select the missed calls menu and highlight the number +1972xxxxxxx and use Dial softkey to dial back to PSTN. This call should use SiteB gateway first. If SB gateway isnt available then it should be routed via HQ gateway. Once the call is connected it should show TO 3xxxxxxxon SB Phone 1 display and From 303XXXX on PSTN Phone Display. When a call goes through HQ, called ID should be 10 digits.

Solution:
R2#debug isdn q931 R2#term moni Nov 25 15:33:54.576: %C3800_ENVM-3-MFAIL_OFF: There is more than one failure with the Power System 1 or this Power System has been turned off. Nov 25 15:33:55.036: ISDN Se0/0/0:23 Q931: RX <- SETUP pd = 8 callref = 0x009B Bearer Capability i = 0x8090A2 Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Progress Ind i = 0x8583 - Origination address is non-ISDN Display i = 'SiteB PSTN' Calling Party Number i = 0x4180, '9723731111' Plan:ISDN, Type:National Called Party Number i = 0xA1, '9723033001' Plan:ISDN, Type:National Do the incoming calling party settings in gw.

Create RL and Route Pattern to route +dialing number from missed call.

/* no need to update the calling party number manipulation done in RL detail level to calling phone so it will display in calling phone as 3xxxxxx. Send the number to calling phone as it is sent by the CUCM */ R2# voice service voip no supplementary-service h225-notify cid-update !

1) MGCP (3 Points)
-- 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 KINDLY NOTE :- There is 2 variations for MGCP Troubleshooting please see the First Event

2) MGCP TROUBLESHOOTING (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. Solution: 1 MGCP GW Registration ccm-manager switchback immediate ccm-manager redundant-host 142.100.64.11 ccm-manager mgcp ! mgcp mgcp call-agent 142.100.64.12 service-type mgcp version 0.1 mgcp dtmf-relay voip codec all mode out-of-band mgcp bind control source-interface GigabitEthernet0/1.102 mgcp bind media source-interface GigabitEthernet0/1.102 Add the gw to CUCM as mgcp gw. Verify the gw registration R1# sh ccm MGCP Domain Name: R1.ccievoice.com Priority Status Host ========================================================= Primary Registered 142.100.64.12 First Backup Backup Ready 142.100.64.11 Second Backup None Current active Call Manager: 142.100.64.12 Backhaul/Redundant link port: 2428

Solution: 2 R1# debug mgcp packets Event1: Backup call-agent sends the message to gw to check the status of active call for port 1 Nov 25 16:36:28.674: MGCP Packet received from 142.100.64.11:2427---> AUEP 24 S0/SU0/DS1-0/1@R1.ccievoice.com MGCP 0.1 F: X, A, I <--Event2: Gateway sends the status message of active calls to the backup call-agent for port 1 Note: Nothing mentioned in I: so it refers no call present in this port Nov 25 16:36:28.674: MGCP Packet sent to 142.100.64.11:2427---> 200 24 I: X: 1 L: p:10-20, a:PCMU;PCMA;G.nX64, b:64, e:on, gc:1, s:on, t:10, r:g, nt:IN;ATM;LOCAL, v:T;G;D;L;H;ATM;FXR L: p:10-220, a:G.729;G.729a;G.729b, b:8, e:on, gc:1, s:on, t:10, r:g, nt:IN;ATM;LOCAL, v:T;G;D;L;H;ATM;FXR L: p:10-110, a:G.726-16;G.728, b:16, e:on, gc:1, s:on, t:10, r:g, nt:IN;ATM;LOCAL, v:T;G;D;L;H;ATM;FXR L: p:10-70, a:G.726-24, b:24, e:on, gc:1, s:on, t:10, r:g, nt:IN;ATM;LOCAL, v:T;G;D;L;H;ATM;FXR L: p:30-330, a:G.723.1-L;G.723.1a-L, b:5, e:on, gc:1, s:on, t:10, r:g, nt:IN;ATM;LOCAL, v:T;G;D;L;H;ATM;FXR M: sendonly, recvonly, sendrecv, inactive, loopback, conttest, data, netwloop, netwtest <--Backup call-agent sends the message to gw to check the status of active call for port 23 Nov 25 16:36:28.698: MGCP Packet received from 142.100.64.11:2427---> AUEP 46 S0/SU0/DS1-0/23@R1.ccievoice.com MGCP 0.1 F: X, A, I <--Gateway sends the status message of active calls to the backup call-agent for port 23 Note: It is sending information I:F so it refers there is active call present in this port Nov 25 16:36:28.698: MGCP Packet sent to 142.100.64.11:2427---> 200 46 I: F X: 17 L: p:10-20, a:PCMU;PCMA;G.nX64, b:64, e:on, gc:1, s:on, t:10, r:g, nt:IN;ATM;LOCAL, v:T;G;D;L;H;ATM;FXR L: p:10-220, a:G.729;G.729a;G.729b, b:8, e:on, gc:1, s:on, t:10, r:g, nt:IN;ATM;LOCAL, v:T;G;D;L;H;ATM;FXR L: p:10-110, a:G.726-16;G.728, b:16, e:on, gc:1, s:on, t:10, r:g, nt:IN;ATM;LOCAL, v:T;G;D;L;H;ATM;FXR L: p:10-70, a:G.726-24, b:24, e:on, gc:1, s:on, t:10, r:g, nt:IN;ATM;LOCAL, v:T;G;D;L;H;ATM;FXR

L: p:10-50, a:G.726-32, b:32, e:on, gc:1, s:on, t:10, r:g, nt:IN;ATM;LOCAL, v:T;G;D;L;H;ATM;FXR L: p:30-270, a:G.723.1-H;G.723;G.723.1a-H, b:6, e:on, gc:1, s:on, t:10, r:g, nt:IN;ATM;LOCAL, v:T;G;D;L;H;ATM;FXR L: p:30-330, a:G.723.1-L;G.723.1a-L, b:5, e:on, gc:1, s:on, t:10, r:g, nt:IN;ATM;LOCAL, v:T;G;D;L;H;ATM;FXR M: sendonly, recvonly, sendrecv, inactive, loopback, conttest, data, netwloop, netwtest <--Event3: Backup call-agent requesting additional information about the active call on port 23 Nov 25 16:36:28.698: MGCP Packet received from 142.100.64.11:2427---> 200 570260852 <--Nov 25 16:36:28.698: MGCP Packet received from 142.100.64.11:2427---> AUCX 47 S0/SU0/DS1-0/23@R1.ccievoice.com MGCP 0.1 I: F F: C, M <---

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. -- 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 fullyglobalized 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.
Solution: R1# gatekeeper zone local GK ccievoice.com 142.1.64.254 no shutdown R3# ephone-dn 1 octo-line number 4001 no-reg primary

! ephone-dn 2 octo-line number 4002 no-reg primary ! ephone-dn 3 number A5001 no-reg primary intercom A5002 label "SC PH2" ! ephone-dn 4 number A5002 no-reg primary intercom A5001 barge-in label "SC PH1 ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip ! interface Loopback0 ip address 142.1.66.254 255.255.255.255 ip ospf network point-to-point h323-gateway voip interface h323-gateway voip id GK ipaddr 142.1.64.254 1719 h323-gateway voip h323-id UCME h323-gateway voip tech-prefix 852 ! gateway ! voice translation-rule 852 rule 1 /^4...$/ /+8522404\0/ ! voice translation-rule 23000 rule 1 /^2...$/ /1408202\0/ type any international plan any isdn rule 2 /^3...$/ /1972303\0/ type any international plan any isdn ! voice translation-rule 852 rule 1 /^85224044/ /4/ ! voice translation-profile from-gk translate called 852 ! voice translation-profile hq-sb translate calling 852 translate called 23000 ! dial-peer voice 852 voip translation-profile incoming from-gk incoming called-number 852. ! dial-peer voice 2300 voip destination-pattern [23]... session target ras tech-prefix 1# dtmf-relay h245-alphanumeric !

dial-peer voice 2301 pots translation-profile outgoing hq-sb preference 1 destination-pattern [23]... port 0/0/0:15 ! CUCM

To register with gk using port number 1720

To use g729 codec from SiteC to CUCM call sitec h323 gw should request the bandwidth from gk. Refer CODEC section of the lab for codec between the sites should be g729.

Add the RL with Local GW of the phone as backup gateway

No digit manipulation in RP calling party and called party number manipulation is done in Route List detail level. Called party manipulation is done in siteb gateway also.

Called party digit manipulation on SiteB gateway R2# ! voice translation-rule 852 rule 1 /^852/ /011852/ type any international plan any isdn ! voice translation-rule 1 rule 1 /^1/ /+1/ ! voice translation-profile to-sitec translate calling 1 translate called 852 ! dial-peer voice 852 pots translation-profile outgoing to-sitec destination-pattern 85224044...$ port 0/0/0:23 ! Verification R1#sh gatek end GATEKEEPER ENDPOINT REGISTRATION ================================ CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags --------------- ----- --------------- ----- ------------ ----142.100.64.11 1720 142.100.64.11 32814 GK VOIP-GW H323-ID: gk-trunk_1 Voice Capacity Max.= Avail.= Current.= 0 142.100.64.12 1720 142.100.64.12 32827 GK VOIP-GW H323-ID: gk-trunk_2 Voice Capacity Max.= Avail.= Current.= 0 142.1.66.254 1720 142.1.66.254 54778 GK H323-GW H323-ID: UCME Voice Capacity Max.= Avail.= Current.= 0 Total number of active registrations = 3 R1#sh gatekeeper gw-type-prefix GATEWAY TYPE PREFIX TABLE ========================= Prefix: 1#* Zone GK master gateway list: 142.100.64.11:1720 gk-trunk_1 142.100.64.12:1720 gk-trunk_2 Prefix: 852* Zone GK master gateway list: 142.1.66.254:1720 UCME

Call from HQ phone to SiteC Phone R1#sh gatekeeper call Total number of active calls = 1. GATEKEEPER CALL INFO ==================== LocalCallID Age(secs) BW 1-29 7 16(Kbps) Endpt(s): Alias E.164Addr src EP: gk-trunk_2 2001 CallSignalAddr Port RASSignalAddr Port 142.100.64.12 1720 142.100.64.12 32827 Endpt(s): Alias E.164Addr dst EP: UCME 85224044001 CallSignalAddr Port RASSignalAddr Port 142.1.66.254 1720 142.1.66.254 54778 Call from SiteC to HQ phone Phone R1#sh gatek call Total number of active calls = 1. GATEKEEPER CALL INFO ==================== LocalCallID Age(secs) BW 2-40639 14 16(Kbps) Endpt(s): Alias E.164Addr src EP: UCME 4001 CallSignalAddr Port RASSignalAddr Port 142.1.66.254 1720 142.1.66.254 54778 Endpt(s): Alias E.164Addr dst EP: gk-trunk_1 1#2001 CallSignalAddr Port RASSignalAddr Port 142.100.64.11 1720 142.100.64.11 32814

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.

Solution: R1# gatekeeper zone local GK ccievoice.com 142.1.64.254 zone local CUBE ccievoice.com zone remote BBGK cisco.com 157.26.1.100 1719 outvia CUBE zone prefix BBGK 01132* no shutdown ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip ! interface GigabitEthernet0/0.102 encapsulation dot1Q 102 ip address 142.102.64.254 255.255.255.0 ip helper-address 142.100.64.11 h323-gateway voip interface h323-gateway voip id CUBE ipaddr 142.1.64.254 1719 h323-gateway voip h323-id HQ ! gateway ! dial-peer voice 1132 voip incoming called-number 01132 ! dial-peer voice 32 voip destination-pattern .T session target ras dtmf-relay h245-alphanumeric ! CUCM

Verification: R1#sh gatek end GATEKEEPER ENDPOINT REGISTRATION ================================ CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags --------------- ----- --------------- ----- ------------ ----142.100.64.11 1720 142.100.64.11 32814 GK VOIP-GW H323-ID: gk-trunk_1 Voice Capacity Max.= Avail.= Current.= 0 142.100.64.12 1720 142.100.64.12 32827 GK VOIP-GW H323-ID: gk-trunk_2 Voice Capacity Max.= Avail.= Current.= 0 142.1.66.254 1720 142.1.66.254 54778 GK H323-GW H323-ID: UCME Voice Capacity Max.= Avail.= Current.= 0 142.102.64.254 1720 142.102.64.254 52364 CUBE H323-GW H323-ID: HQ Voice Capacity Max.= Avail.= Current.= 0 Total number of active registrations = 4

R1#sh gatekeeper gw-type-prefix GATEWAY TYPE PREFIX TABLE ========================= Prefix: 1#* Zone GK master gateway list: 142.100.64.11:1720 gk-trunk_1 142.100.64.12:1720 gk-trunk_2 Prefix: 852* Zone GK master gateway list: 142.1.66.254:1720 UCME Note: CUBE is not required to register to gk with zone prefix or tech-prefix. Verify the call going through CUBE. R1#sh voip rtp connection detail R1#sh gatek call Total number of active calls = 2. GATEKEEPER CALL INFO ==================== LocalCallID Age(secs) BW 64-238 4 16(Kbps) Endpt(s): Alias E.164Addr src EP: gk-trunk_2 14082022001 CallSignalAddr Port RASSignalAddr Port 142.100.64.12 1720 142.100.64.12 32827 Endpt(s): Alias E.164Addr dst EP: HQ 0113212345555 CallSignalAddr Port RASSignalAddr Port 142.102.64.254 1720 142.102.64.254 52364 LocalCallID Age(secs) BW 65-238 4 16(Kbps) Endpt(s): Alias E.164Addr src EP: HQ 14082022001 CallSignalAddr Port RASSignalAddr Port 142.102.64.254 1720 142.102.64.254 52364 Endpt(s): Alias E.164Addr dst EP: 0113212345555 CallSignalAddr Port RASSignalAddr Port 157.26.1.100 1720 157.26.1.100 1720 R1#

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.

Solution: R1#debug cch323 h225 /* to troubleshoot the issue with CUBE */ debug cch323 h225 debug cch323 h245 1. The remote phone is picking up and the "H225 connect" message is being sent back to the CUBE. ***Jun 5 21:09:58.858: //-1/xxxxxxxxxxxx/H323/cch323_h225_receiver: Received msg of type SETUPCFM_CHOSEN 2. The CUBE by default is not sending the H225 connect message back to the HQ phone because it is waiting for the H245 negotiation to take place. ***Jun 10 21:09:58.858: //100/80C2A9AE1800/H323/cch323_h225_set_new_state: Changing from H225_ALERT state to H225_WAIT_FOR_H245 state ***Jun 5 21:09:58.858: //99/80C2A9AE1800/H323/cch323_h245_set_new_state: Changing from H245_NONE state to H245_WAITING state 3. The HQ phone is waiting for the H225 connect message to begin the H245 negotiation, but has not received it because the CUBE by default waits for the H245 negotiation. = 157.26.1.100 Nov 26 04:50:12.710: //95/80D1959E0800/H323/run_h225_sm: Received event H225_EV_RELEASE while at state H225_ALERT Nov 26 04:50:12.710: //95/80D1959E0800/H323/cch323_h225_set_new_state: Changing from H225_ALERT state to H225_WAIT_FOR_DRQ state Nov 26 04:50:12.710: //95/80D1959E0800/H323/cch323_h225_send_release: Cause = 47; Location = 0 Nov 26 04:50:12.710: //95/80D1959E0800/H323/cch323_h225_send_release: h225TerminateRequest: src address = -1906032386; dest address = 142.100.64.12 Nov 26 04:50:12.710: //95/80D1959E0800/H323/run_h225_sm: Received event H225_EV_H245_CONN_LOST while at state H225_WAIT_FOR_DRQ Nov 26 04:50:12.714: //95/80D1959E0800/H323/run_h225_sm: Received event H225_EV_CONN_LOST while at state H225_WAIT_FOR_DRQ So change the gk-trunk option not to wait for Far End H.245 Terminal Capability Set

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. Solution: Create Regions and assign to Device Pools accordingly

Call from HQ phone to SiteC Phone R1#sh gatekeeper call Total number of active calls = 1. GATEKEEPER CALL INFO ==================== LocalCallID Age(secs) BW 1-29 7 16(Kbps) Endpt(s): Alias E.164Addr src EP: gk-trunk_2 2001 CallSignalAddr Port RASSignalAddr Port 142.100.64.12 1720 142.100.64.12 32827 Endpt(s): Alias E.164Addr dst EP: UCME 85224044001 CallSignalAddr Port RASSignalAddr Port 142.1.66.254 1720 142.1.66.254 54778

Call from SiteC to HQ phone Phone R1#sh gatek call Total number of active calls = 1. GATEKEEPER CALL INFO ==================== LocalCallID Age(secs) BW 2-40639 14 16(Kbps) Endpt(s): Alias E.164Addr src EP: UCME 4001 CallSignalAddr Port RASSignalAddr Port 142.1.66.254 1720 142.1.66.254 54778 Endpt(s): Alias E.164Addr dst EP: gk-trunk_1 1#2001 CallSignalAddr Port RASSignalAddr Port 142.100.64.11 1720 142.100.64.11 32814

7. Media Resource Management

Cbarge - (3 points) 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 Solution: Note: If cBarge has to work in SRST, Phone configuration in CUCM should be Single button barge to Default, Bulit-in Bridge off and Privacy off. R2# sccp local Vlan302 sccp ccm 142.100.64.11 identifier 2 version 7.0 sccp ccm 142.100.64.12 identifier 1 version 7.0 sccp ! sccp ccm group 1 bind interface Vlan302 associate ccm 1 priority 1 associate ccm 2 priority 2 associate profile 1 register sb-cfb ! dspfarm profile 1 conference codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 3 associate application SCCP

If required create Softkey Template and assign the template to phone.

Assign the Conference Bridge to MRG, assign the MRG to MRGL and assign the MRGL to SiteB Device Pool and reset the DP. SiteB Ph1

If they asked not to use softkey template change the single button barge to cBarge

Do the same settings for SiteB Ph2 also. Verification: make inbound call to 3012 answer in any of the phone and press the shared line on the other phone and verify the below result. R2# sh sccp connection sess_id conn_id stype mode 33555433 33555433 33555433 33554452 33554450 33554448 codec ripaddr rport sport

conf sendrecv g711u conf sendrecv g711u conf sendrecv g711u

142.102.65.254 16834 17808 142.102.65.29 17956 17158 142.102.65.30 22312 18136

Total number of active session(s) 1, and connection(s) 3

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. If the call is not retrieved it has to try to redeliver in 45 secs. This feature should be available only for HQ ph2.

Solution: This should be available to hq ph2 only so create the new partition to associate with this number. Create the separate css for hq ph2 and include this partition into the css and assign the new css to hq ph2.

Create Phone button Template

Final Output of the phone

Testing: Call to 2002 from any phone transfer to 2802 then press transfer key again. Now try to retrieve the call from 2001 dial *2802 it will fail now try from 2002 call will get connected.

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
Solution: R3# ephone-dn 5 octo-line number 4321 no-reg primary conference meetme ! ephone-template 1 conference drop-mode local softkeys idle Redial Newcall Cfwdall Pickup Dnd softkeys seized Pickup Cfwdall Endcall Redial Meetme softkeys connected Hold Endcall Trnsfer Confrn Park ! ephone 1 ephone-template 1 ! ephone 2 ephone-template 1 ! voice class custom-cptone leave dualtone conference frequency 300 cadence 300 250 ! voice class custom-cptone Join dualtone conference frequency 700 cadence 300 50 300 50 !

sccp local GigabitEthernet0/1.502 sccp ccm 142.1.66.254 identifier 1 version 7.0 sccp ! sccp ccm group 1 bind interface GigabitEthernet0/1.502 associate ccm 1 priority 1 associate profile 2 register sc-cfb ! dspfarm profile 2 conference codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 3 conference-join custom-cptone Join conference-leave custom-cptone leave associate application SCCP ! telephony-service sdspfarm units 2 sdspfarm tag 2 sc-cfb conference hardware max-conferences 12 gain -6 ! Verification: Make call to 4321 from any sitec phone, call from pstn phone to 24044321, call from other sitec phone and verify the result. R3#sh sccp conn sess_id conn_id stype mode codec ripaddr rport sport 2000 16594 2000 19536 2000 18300

3221553157 131085 3221553157 131084 3221553157 131083

conf sendrecv g711u conf sendrecv g711u conf sendrecv g711u

142.1.66.254 142.1.66.254 142.1.66.254

Total number of active session(s) 1, and connection(s) 3

8. QoS 8.1 Switch QoS LAN QOS - 5 points

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

Solution: interface GigabitEthernet0/5 auto qos voip cisco-phone mls mls mls mls mls qos qos qos qos qos srr-queue srr-queue srr-queue srr-queue srr-queue output output output output output cos-map cos-map cos-map cos-map cos-map queue queue queue queue queue 1 2 2 3 4 threshold threshold threshold threshold threshold 3 3 2 3 3 5 67 4 23 01

interface GigabitEthernet0/5 srr-queue bandwidth share 4 30 40 30 srr-queue bandwidth shape 4 0 0 0 queue-set 2 interface is configured for queue-set 2 so we have to change the drop threshold for queueset 2 and queue 2. mls qos queue-set output 2 threshold 2 60 60 50 200 Verification: HQ-SW#sh mls qos maps cos-output-q Cos-outputq-threshold map: cos: 0 1 2 3 4 5 6 7 -----------------------------------queue-threshold: 4-3 4-3 3-3 3-3 2-2 1-3 2-3 2-3 HQ-SW#sh mls qos interface g0/5 queueing GigabitEthernet0/5 Egress Priority Queue : enabled Shaped queue weights (absolute) : 4 0 0 0 Shared queue weights : 4 30 40 30 The port bandwidth limit : 100 (Operational Bandwidth:100.0) The port is mapped to qset : 2 HQ-SW#sh mls qos queue-set 2 Queueset: 2 Queue : 1 2 3 4 ---------------------------------------------buffers : 16 6 17 61 threshold1: 149 60 100 42 threshold2: 149 200 100 72 reserved : 100 50 50 100 maximum : 149 200 400 242

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)

Solution: HQ - SiteB: R1# ! interface Serial0/0/1:0.201 point-to-point description ** T1 DATA CONNECTION to BR1 ** bandwidth 384 ip address 142.1.67.1 255.255.255.252 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 201 class AutoQoS-FR-Se0/0/1:0-201 auto qos voip trust frame-relay ip rtp header-compression //* Remove Compression if not asked to do *// ! interface Serial0/0/1:0 /* Physical interface */ encapsulation frame-relay frame-relay traffic-shaping frame-relay lmi-type ansi ! class-map match-any AutoQoS-VoIP-RTP-Trust match ip dscp ef class-map match-any AutoQoS-VoIP-Control-Trust match ip dscp cs3 match ip dscp af31 ! policy-map AutoQoS-Policy-Trust class AutoQoS-VoIP-RTP-Trust priority percent 33 class AutoQoS-VoIP-Control-Trust bandwidth percent 5 class class-default fair-queue ! map-class frame-relay AutoQoS-FR-Se0/0/1:0-201 frame-relay cir 364800 frame-relay bc 3648 frame-relay be 0 frame-relay mincir 364800 //* 95% QoS SRND Recommendation *// frame-relay fragment 480 service-policy output AutoQoS-Policy-Trust ! Note: Do the same configuration in SiteB (R2) router also.

HQ-SiteC: R1# ! interface Serial0/0/1:0.202 point-to-point description ** T1 DATA CONNECTION to BR2 ** bandwidth 768 ip address 142.1.67.5 255.255.255.252 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 202 class AutoQoS-FR-Se0/0/1:0-202 auto qos voip trust frame-relay ip rtp header-compression //* Remove Compression if not asked to do *// ! class-map match-any AutoQoS-VoIP-RTP-Trust match ip dscp ef class-map match-any AutoQoS-VoIP-Control-Trust match ip dscp cs3 match ip dscp af31 ! policy-map AutoQoS-Policy-Trust class AutoQoS-VoIP-RTP-Trust priority percent 33 class AutoQoS-VoIP-Control-Trust bandwidth percent 5 class class-default fair-queue ! map-class frame-relay AutoQoS-FR-Se0/0/1:0-202 frame-relay cir 729600 frame-relay bc 7296 frame-relay be 0 frame-relay mincir 729600 frame-relay fragment 960 service-policy output AutoQoS-Policy-Trust Verification: R1# sh frame-relay pvc 202 PVC Statistics for interface Serial0/0/1:0 (Frame Relay DTE) DLCI = 201, DLCI USAGE = LOCAL, PVC STATUS = ACTIVE, INTERFACE = Serial0/0/1:0.201 fragment type end-to-end fragment size 480 cir 364800 bc 3648 be 0 byte limit 456 interval 10 mincir 364800 byte increment 456 BECN response no IF_CONG no

R1#sh traffic-shape Interface Se0/0/1:0.201 Access Target Byte Sustain Excess Interval Increment Adapt VC List Rate Limit bits/int bits/int (ms) (bytes) Active 201 729600 912 7296 0 10 912 Interface Se0/0/1:0.202 Access Target Byte Sustain Excess Interval Increment Adapt VC List Rate Limit bits/int bits/int (ms) (bytes) Active 202 384000 480 3840 0 10 480 R1# sh policy-map interface Serial0/0/1:0.201: DLCI 201 Service-policy output: AutoQoS-Policy-Trust queue stats for all priority classes: queue limit 64 packets (queue depth/total drops/no-buffer drops) 0/0/0 (pkts output/bytes output) 0/0 Class-map: AutoQoS-VoIP-RTP-Trust (match-any) 1138 packets, 72832 bytes 5 minute offered rate 4000 bps, drop rate 0 bps Match: ip dscp ef (46) 1138 packets, 72832 bytes 5 minute rate 4000 bps Priority: 33% (120 kbps), burst bytes 3000, b/w exceed drops: 0

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 246810 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 Solution: CUCM Configuration: Change the AXL (Administrator) user permission

Update the CSS and partition of the voice mail port, MWI numbers, voice Mail Pilot, Voice Mail Profile and Hunt Pilot Settings.

Create the end users (hq ph1, siteb ph1, siteb ph2) to be associated with mail boxes and assign to the phones. Configure the primary extension.

Configure the DN settings to forward the calls to voicemail system.

Unity Connection Configuration Start all services in Unity Connection

Verify Phone System Settings

Edit AXL server settings and user credential

Verify Port Group Settings

Verify the Ports

Edit User Template Settings

Change the password

Import the Users from CUCM

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 GUI web administrator: administrator GUI web password: ccievoice Solution: CME Router Configuration R3(config)# R3(config)#interface integrated service-Engine 1/0 R3(config-if)#ip unnumbered vlan 502 R3(config-if)#service-module ip address 142.102.66.253 255.255.255.0 R3(config-if)#service-module ip default-gateway 142.102.66.254 R3(config-if)#no shut R3(config)#ip route 142.102.66.253 255.255.255.255 integrated service-Engine 1/0

CUE Configuration R3#service-module integrated service-Engine 1/0 session CUE# offline /* If required */ CUE(offline)# restore factory default /* If Required */ Do you wish to start configuration now (y,n)? y Are you sure (y,n)? y Enter Hostname (my-hostname, or enter to use se-142-102-66-253): CUE Enter Domain Name (mydomain.com, or enter to use localdomain): ccievoice.com Would you like to use DNS (y,n)?n WARNING: If DNS is not used, IP addresses will be required. Are you sure (y,n)? y Enter IP Address of the Primary NTP Server (IP address, or enter for 142.102.66.254): 142.102.64.254 Found server 142.102.64.254 Enter IP Address of the Secondary NTP Server (IP address, or enter to bypass): Please identify a location so that time zone rules can be set correctly. Please select a continent or ocean. 1) Africa 4) Arctic Ocean 7) Australia 10) Pacific Ocean 2) Americas 5) Asia 8) Europe 3) Antarctica 6) Atlantic Ocean 9) Indian Ocean #? 5 Please select a country. 13) Hong Kong 14) India #? 13 30) Mongolia 31) Myanmar (Burma) 47) United Arab Emirates 48) Uzbekistan

Is the above information OK? 1) Yes 2) No #? 1 Enter administrator user ID: (user ID): administrator

Enter password for administrator: (password): ccievoice Confirm password for administrator by reentering it: (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.
-- 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 Solution: CUCME configuration: R3# ! interface integrated Service-Engine1/0 ip unnumbered GigabitEthernet0/1.502 service-module ip address 142.102.66.253 255.255.255.0 service-module ip default-gateway 142.102.66.254 no shut ! ip route 142.102.66.253 255.255.255.255 Service-Engine1/0 !

ip http server ip http path flash:gui/ ip http authentication local ! telephony-service max-dn 5 no-reg both voicemail 4220 web admin system name admin password cisco dn-webedit ! ephone-dn 3 number 1998.... no-reg mwi on ! ephone-dn 4 number 1999.... no-reg mwi off ! sip-ua mwi-server ipv4:142.102.66.253 ! dial-peer voice 4220 voip destination-pattern 4220 session protocol sipv2 session target ipv4:142.102.66.253 incoming called-number 199[89].... dtmf-relay sip-notify codec g711ulaw no vad ! ephone 1 mac-address 001B.D4A0.9428 username "sitecph1" ! ephone 2 mac-address 001B.D4C6.D8B6 username "sitecph2" ! sccp local loopback0 sccp ccm 142.1.66.254 identifier 1 version 7.0 sccp ! sccp ccm group 1 bind interface vlan 502 associate ccm 1 priority 1 associate profile 1 register sc-xcode associate profile 2 register sc-cfb !

dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 maximum sessions 3 associate application SCCP ! CUE configuration: Verify that the required license is available to integrate with CME. If not, install the license from available source. #service-module integrated service-engine 1/0 session #show software license #software install clean url ftp://serverip/ cue-vm-license_12mbx_cme_7.0.1.pkg username cisco password cisco

Modify the user settings like PIN, no Tutorial and etc.

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. Solution:

Do the following from CUE CLI CUE(config)# voicemail callerid

10: UCCX Applications (5 points)

Create the following script -- 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. Solution:

11: 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 2 as shown in the picture.

Solution: R3(config)#sip-ua R3(config-sip-ua)#presence enable R3(config)#presence R3(config-presence)#presence call-list R3(config)#ephone-dn 1 R3(config-ephone-dn)#allow watch R3(config)#ephone-dn 2 R3(config-ephone-dn)#allow watch R3(config)#ephone 1 R3(config-ephone)#presence call-list R3(config)#ephone 2 R3(config-ephone)#presence call-list R3(config-ephone)#button 1:2 2:4 3m1 R3(config)#telephony-service R3(config-telephony)#directory entry 1 4001 name Site C Phone 1 R3(config-telephony)#directory entry 2 4002 name Site C Phone 2

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). Solution:

R2# ! ephone-dn-template 1 call-forward busy 914082022220 call-forward noan 914082022220 timeout 20 ! telephony-service srst mode auto-provision dn Note: If CBarge has to work in SRST use auto-provision all. srst dn template 1 srst dn line-mode octo max-ephones 5 max-dn 5 preference 9 ip source-address 142.102.65.254 port 2000 voicemail 914082022220 time-zone 8 time-format 24 date-format mm-dd-yy max-conferences 12 gain -6 transfer-system full-consult create cnf-files version-stamp 7960 Nov 26 2010 10:59:30 ! ephone-dn 1 octo-line number 3002 description +19723033002 name SiteB Ph2 preference 9 ephone-dn-template 1 ! ephone-dn 2 octo-line number 3012 description 3012 name Shared Line preference 9 ephone-dn-template 1 !

ephone-dn 3 octo-line number 3001 description +19723033001 name SiteB Ph1 preference 9 ephone-dn-template 1 ! voice translation-rule 11 rule 1 /^3...$/ /+1972303\0/ ! voice translation-rule 9011 rule 1 // // type any international plan any isdn ! voice translation-rule 24000 rule 1 /^2...$/ /1408202\0/ type any national plan any isdn rule 2 /^4...$/ /0118522404\0/ type any international plan any isdn ! voice translation-profile srst translate calling 11 translate called 24000 ! voice translation-profile international translate calling 11 translate called 9011 ! dial-peer voice 24000 pots translation-profile outgoing srst destination-pattern [24]...$ port 0/0/0:23 ! dial-peer voice 2220 pots destination-pattern 914082022220 forward-digits 11 port 0/0/0:23 ! dial-peer voice 9011 pots translation-profile outgoing international destination-pattern 9011T port 0/0/0:23 prefix 011 ! dial-peer hunt 2 ! Note: Do the RDNIS in HQ gw and SiteB GW for unity connection to play users personal greeting to the calling person.

SRST ADVANCE
-- 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 -- When in SRST phone it should still be able to utilize Cbarge -- 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. Solution: Do the same as previous section and additionally add the conference configuration. If they ask to display 4 digit ani on hq phone and sitec phones dont user voice translation profiles to change the 4-digit into e164 number. telephony-service sdspfarm units 2 sdspfarm tag 1 sb-cfb conference hardware srst mode auto-provision all Note: If CBarge has to work in SRST use auto-provision all. srst ephone template 1 srst dn template 1 srst dn line-mode octo max-ephones 5 max-dn 5 preference 9 time-zone 40 ip source-address 142.102.65.254 port 2000 voicemail 2220 max-conferences 12 gain -6 transfer-system full-consult create cnf-files version-stamp 7960 Nov 26 2010 10:59:30 ! ephone-dn-template 1 call-forward busy 2220 call-forward noan 2220 timeout 20 ! ephone-template 1 privacy off conference drop-mode local softkeys remote-in-use Newcall CBarge

softkeys idle Redial Newcall Cfwdall softkeys seized Cfwdall Endcall Meetme Pickup Redial softkeys connected Confrn Endcall Join Park Select Trnsfer ! ephone-dn 4 octo-line number 3111 conference ad-hoc ! sccp local Vlan302 sccp ccm 142.102.65.254 identifier 3 version 6.0 sccp ccm 142.100.64.11 identifier 2 version 6.0 sccp ccm 142.100.64.12 identifier 1 version 6.0 sccp ! sccp ccm group 1 bind interface Vlan302 associate ccm 1 priority 1 associate ccm 2 priority 2 associate ccm 3 priority 3 associate profile 1 register sb-cfb ! dspfarm profile 1 conference maximum sessions 3 associate application SCCP ! voice translation-rule 23 rule 1 /^2...$/ /1408202\0/ type any national plan any isdn ! voice translation-profile hq translate called 23 ! dial-peer voice 2000 pots translation-profile outgoing hq destination-pattern 2... no digit-strip port 0/0/0:23 !

Final configuration of Routers

HQ Router: hostname R1 ! clock timezone PST -8 clock summer-time PST recurring network-clock-participate wic 0 ! no ip domain lookup ip domain name ccievoice.com no ipv6 cef ! isdn switch-type primary-ni ! voice-card 0 no dspfarm ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip ! controller T1 0/0/0 framing esf linecode b8zs pri-group timeslots 1-24 service mgcp ! controller T1 0/0/1 framing esf linecode b8zs channel-group 0 timeslots 1-24 description ** T1 DATA CONNECTION to PSTN-RTR ** ! class-map match-any AutoQoS-VoIP-RTP-Trust match ip dscp ef class-map match-any AutoQoS-VoIP-Control-Trust match ip dscp cs3 match ip dscp af31 ! policy-map AutoQoS-Policy-Trust class AutoQoS-VoIP-RTP-Trust priority percent 33 class AutoQoS-VoIP-Control-Trust bandwidth percent 5 class class-default

fair-queue ! interface Loopback0 ip address 142.1.64.254 255.255.255.255 ip ospf network point-to-point ! interface GigabitEthernet0/1 duplex auto speed auto media-type rj45 ! interface GigabitEthernet0/1.100 encapsulation dot1Q 100 ip address 142.100.64.254 255.255.255.0 ! interface GigabitEthernet0/1.102 encapsulation dot1Q 102 ip address 142.102.64.254 255.255.255.0 ip helper-address 142.100.64.11 h323-gateway voip interface h323-gateway voip id CUBE ipaddr 142.1.64.254 1719 h323-gateway voip h323-id HQ ! interface GigabitEthernet0/1.202 encapsulation dot1Q 202 ip address 142.202.64.254 255.255.255.0 ! interface Serial0/0/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice voice isdn bind-l3 ccm-manager no cdp enable ! interface Serial0/0/1:0 no ip address encapsulation frame-relay frame-relay traffic-shaping frame-relay lmi-type ansi ! interface Serial0/0/1:0.201 point-to-point description ** T1 DATA CONNECTION to BR1 ** bandwidth 768 ip address 142.1.67.1 255.255.255.252 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 201

class AutoQoS-FR-Se0/0/1:0-201 auto qos voip trust frame-relay ip rtp header-compression ! interface Serial0/0/1:0.202 point-to-point description ** T1 DATA CONNECTION to BR2 ** bandwidth 384 ip address 142.1.67.5 255.255.255.252 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 202 class AutoQoS-FR-Se0/0/1:0-202 auto qos voip trust frame-relay ip rtp header-compression ! map-class frame-relay AutoQoS-FR-Se0/0/1:0-201 frame-relay cir 729600 frame-relay bc 7296 frame-relay be 0 frame-relay mincir 729600 frame-relay fragment 960 service-policy output AutoQoS-Policy-Trust ! map-class frame-relay AutoQoS-FR-Se0/0/1:0-202 frame-relay cir 384000 frame-relay bc 3840 frame-relay be 0 frame-relay mincir 384000 frame-relay fragment 480 service-policy output AutoQoS-Policy-Trust ! voice-port 0/0/0:23 ! ccm-manager switchback immediate ccm-manager redundant-host 142.100.64.11 ccm-manager mgcp ! mgcp mgcp call-agent 142.100.64.12 service-type mgcp version 0.1 mgcp dtmf-relay voip codec all mode out-of-band mgcp bind control source-interface GigabitEthernet0/1.102 mgcp bind media source-interface GigabitEthernet0/1.102 ! mgcp profile default ! dial-peer voice 1132 voip incoming called-number 01132 !

dial-peer voice 32 voip destination-pattern .T session target ras dtmf-relay h245-alphanumeric ! gateway timer receive-rtp 1200 ! ! gatekeeper zone local GK ccievoice.com zone local CUBE ccievoice.com zone remote BBGK cisco.com 157.26.1.100 1719 outvia CUBE zone prefix BBGK 01132* no shutdown ! ntp server 157.26.1.100

SiteB Router: hostname R2 ! clock timezone CST -6 clock summer-time CST recurring network-clock-participate wic 0 ! no ip domain lookup ip domain name ccievoice.com ! isdn switch-type primary-ni ! voice-card 0 no dspfarm dsp services dspfarm ! voice service voip no supplementary-service h225-notify cid-update ! voice class codec 1 codec preference 1 g711ulaw bytes 160 codec preference 2 g729r8 bytes 20 ! voice class h323 1 h225 timeout tcp establish 2 ! voice translation-rule 4 rule 1 /^9723033/ /3/

voice translation-rule 7 rule 1 /^3...$/ /303\0/ ! voice translation-rule 9 rule 1 // // type any subscriber plan any isdn ! voice translation-rule 10 rule 1 /^3...$/ /972303\0/ ! voice translation-rule 11 rule 1 /^3...$/ /+1972303\0/ ! voice translation-rule 23 rule 1 /^2...$/ /1408202\0/ type any national plan any isdn ! voice translation-rule 40 rule 1 /^4...$/ /0118522404\0/ type any international plan any isdn ! voice translation-rule 91 rule 1 // // type any national plan any isdn ! voice translation-rule 9011 rule 1 // // type any international plan any isdn ! voice translation-profile 911 translate calling 10 ! voice translation-profile did translate called 4 ! voice translation-profile hq translate calling 11 translate called 23 ! voice translation-profile international translate calling 11 translate called 9011 ! voice translation-profile intl translate calling 9011 translate called 852 ! voice translation-profile local translate calling 7 translate called 9 ! voice translation-profile national translate calling 10

translate called 91 ! voice translation-profile sitec translate calling 11 translate called 40 ! vtp domain ccie vtp mode transparent ! controller T1 0/0/0 framing esf linecode b8zs pri-group timeslots 1-24 ! controller T1 0/0/1 framing esf linecode b8zs channel-group 0 timeslots 1-24 description ** T1 Data Connection to PSTN-RTR ** ! vlan 302 name Voice ! vlan 402 name Data ! class-map match-any AutoQoS-VoIP-RTP-Trust match ip dscp ef class-map match-any AutoQoS-VoIP-Control-Trust match ip dscp cs3 match ip dscp af31 ! policy-map AutoQoS-Policy-Trust class AutoQoS-VoIP-RTP-Trust priority percent 33 class AutoQoS-VoIP-Control-Trust bandwidth percent 5 class class-default fair-queue ! interface Loopback0 ip address 142.1.65.254 255.255.255.255 ip ospf network point-to-point ! interface Serial0/0/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni

isdn incoming-voice voice isdn outgoing display-ie isdn outgoing ie redirecting-number no cdp enable ! interface Serial0/0/1:0 no ip address encapsulation frame-relay frame-relay traffic-shaping frame-relay lmi-type ansi ! interface Serial0/0/1:0.101 point-to-point description ** T1 DATA CONNECTION to HQ-RTR ** bandwidth 768 ip address 142.1.67.2 255.255.255.252 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 101 class AutoQoS-FR-Se0/0/1:0-101 auto qos voip trust frame-relay ip rtp header-compression ! interface FastEthernet1/0 switchport trunk native vlan 402 switchport mode trunk switchport voice vlan 302 spanning-tree portfast ! interface FastEthernet1/1 switchport trunk native vlan 402 switchport mode trunk switchport voice vlan 302 spanning-tree portfast ! interface Vlan302 ip address 142.102.65.254 255.255.255.0 ip helper-address 142.100.64.11 h323-gateway voip interface h323-gateway voip bind srcaddr 142.102.65.254 ! interface Vlan402 ip address 142.202.65.254 255.255.255.0 ! ip forward-protocol nd no ip http server no ip http secure-server !

! map-class frame-relay AutoQoS-FR-Se0/0/1:0-101 frame-relay cir 729600 frame-relay bc 7296 frame-relay be 0 frame-relay mincir 729600 frame-relay fragment 960 service-policy output AutoQoS-Policy-Trust ! voice-port 0/0/0:23 translation-profile incoming did ! sccp local Vlan302 sccp ccm 142.100.64.12 identifier 1 version 6.0 sccp ccm 142.100.64.11 identifier 2 version 6.0 sccp ccm 142.102.65.254 identifier 3 version 6.0 sccp ! sccp ccm group 1 bind interface Vlan302 associate ccm 1 priority 1 associate ccm 2 priority 2 associate ccm 3 priority 3 associate profile 1 register sb-cfb keepalive retries 2 keepalive timeout 3 ! dspfarm profile 1 conference codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 3 associate application SCCP ! dial-peer voice 1 pots incoming called-number . direct-inward-dial port 0/0/0:23 ! dial-peer voice 911 pots translation-profile outgoing 911 destination-pattern 911 no digit-strip port 0/0/0:23 !

dial-peer voice 3000 voip destination-pattern 3...$ voice-class codec 1 voice-class h323 1 session target ipv4:142.100.64.12 incoming called-number . dtmf-relay h245-alphanumeric ! dial-peer voice 3001 voip preference 1 destination-pattern 3...$ voice-class codec 1 voice-class h323 1 session target ipv4:142.100.64.11 dtmf-relay h245-alphanumeric ! dial-peer voice 9 pots translation-profile outgoing local destination-pattern 9[2-9]...... port 0/0/0:23 forward-digits 7 ! dial-peer voice 91 pots translation-profile outgoing national destination-pattern 91.......... port 0/0/0:23 forward-digits 11 ! dial-peer voice 9011 pots translation-profile outgoing international destination-pattern 9011T port 0/0/0:23 prefix 011 ! dial-peer voice 2000 pots translation-profile outgoing hq destination-pattern 2... no digit-strip port 0/0/0:23 ! dial-peer voice 4000 pots translation-profile outgoing sitec destination-pattern 4... no digit-strip port 0/0/0:23 ! dial-peer voice 2220 pots destination-pattern 2220

no digit-strip port 0/0/0:23 prefix 1408202 ! telephony-service sdspfarm units 2 sdspfarm tag 1 sb-cfb conference hardware srst mode auto-provision none srst ephone template 1 srst dn template 1 srst dn line-mode octo max-ephones 5 max-dn 5 preference 9 ip source-address 142.102.65.254 port 2000 time-zone 40 voicemail 2220 max-conferences 12 gain -6 transfer-system full-consult create cnf-files version-stamp Jan 01 2002 00:00:00 ! ephone-dn-template 1 call-forward busy 2220 call-forward noan 2220 timeout 20 ! ephone-template 1 privacy off softkeys remote-in-use CBarge Newcall softkeys idle Redial Newcall softkeys seized Endcall Redial Pickup softkeys connected Endcall Confrn Trnsfer Park ! ephone-dn 4 octo-line number 3111 conference ad-hoc preference 9 ! ntp server 142.1.64.254

SiteC Router: hostname R3 ! clock timezone HKT 8 network-clock-participate wic 0 ! !

! ip dhcp excluded-address 142.102.66.1 142.102.66.9 ip dhcp excluded-address 142.102.66.31 142.102.66.254 ! ip dhcp pool CME network 142.102.66.0 255.255.255.0 default-router 142.102.66.254 option 150 ip 142.1.66.254 ! isdn switch-type primary-net5 ! voice-card 0 no dspfarm dsp services dspfarm ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip ! voice class custom-cptone leave dualtone conference frequency 300 cadence 300 250 ! voice class custom-cptone Join dualtone conference frequency 700 cadence 300 50 300 50 ! voice translation-rule 4 rule 1 /^24044/ /4/ ! voice translation-rule 8 rule 1 /^4...$/ /2404\0/ ! voice translation-rule 9 rule 1 // // type any subscriber plan any isdn ! voice translation-rule 11 rule 1 /^4...$/ /+8522404\0/ ! voice translation-rule 23 rule 1 /^2...$/ /1408202\0/ type any international plan any isdn rule 2 /^3...$/ /1972303\0/ type any international plan any isdn ! voice translation-rule 852

rule 1 /^85224044/ /4/ ! voice translation-rule 900 rule 1 // // type any international plan any isdn ! voice translation-profile 999 translate calling 8 ! voice translation-profile did translate called 4 ! voice translation-profile from-gk translate called 852 ! voice translation-profile hq-sb translate calling 11 translate called 23 ! voice translation-profile international translate calling 11 translate called 900 ! voice translation-profile local translate calling 8 translate called 9 ! controller E1 0/0/0 pri-group timeslots 1-12,16 ! controller E1 0/0/1 channel-group 0 timeslots 1-31 description ** E1 Data Connection to PSTN-RTR ** ! ! class-map match-any AutoQoS-VoIP-RTP-Trust match ip dscp ef class-map match-any AutoQoS-VoIP-Control-Trust match ip dscp cs3 match ip dscp af31 ! policy-map AutoQoS-Policy-Trust class AutoQoS-VoIP-RTP-Trust priority percent 33 class AutoQoS-VoIP-Control-Trust bandwidth percent 5 class class-default fair-queue !

interface Loopback0 ip address 142.1.66.254 255.255.255.255 ip ospf network point-to-point h323-gateway voip interface h323-gateway voip id GK ipaddr 142.1.64.254 1719 h323-gateway voip h323-id UCME h323-gateway voip tech-prefix 852 ! interface vlan 502 ip address 142.102.66.254 255.255.255.0 ! interface vlan 602 ip address 142.202.66.254 255.255.255.0 ! interface Serial0/0/0:15 no ip address encapsulation hdlc isdn switch-type primary-net5 isdn incoming-voice voice isdn outgoing display-ie isdn outgoing ie redirecting-number no cdp enable ! interface Serial0/0/1:0 no ip address encapsulation frame-relay frame-relay traffic-shaping frame-relay lmi-type ansi ! interface Serial0/0/1:0.102 point-to-point description ** E1 DATA CONNECTION to HQ-RTR ** bandwidth 384 ip address 142.1.67.6 255.255.255.252 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 102 class AutoQoS-FR-Se0/0/1:0-102 auto qos voip trust frame-relay ip rtp header-compression ! interface Service-Engine1/0 ip unnumbered GigabitEthernet0/1.502 service-module ip address 142.102.66.253 255.255.255.0 service-module ip default-gateway 142.102.66.254 ! ip forward-protocol nd ip route 142.102.66.253 255.255.255.255 Service-Engine1/0 ip http server

ip http authentication local no ip http secure-server ip http path flash:gui ! map-class frame-relay AutoQoS-FR-Se0/0/1:0-102 frame-relay cir 364800 frame-relay bc 3648 frame-relay be 0 frame-relay mincir 364800 frame-relay fragment 480 service-policy output AutoQoS-Policy-Trust ! tftp-server flash:Desktops/320x196x4/List.xml tftp-server flash:Desktops/320x196x4/ipxlogo.png tftp-server flash:Desktops/320x196x4/ipxlogo-tn.png ! voice-port 0/0/0:15 translation-profile incoming did ! sccp local vlan 502 sccp ccm 142.1.66.254 identifier 1 version 6.0 sccp ! sccp ccm group 1 bind interface vlan 502 associate ccm 1 priority 1 associate profile 1 register sitec-xcode associate profile 2 register sitec-cfb ! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 maximum sessions 3 associate application SCCP ! dspfarm profile 2 conference codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 3 conference-join custom-cptone Join conference-leave custom-cptone leave

associate application SCCP ! ! dial-peer voice 1 pots incoming called-number . direct-inward-dial port 0/0/0:15 ! dial-peer voice 999 pots translation-profile outgoing 999 destination-pattern 999 port 0/0/0:15 forward-digits all ! dial-peer voice 9 pots translation-profile outgoing local destination-pattern 9[2-8]....... port 0/0/0:15 forward-digits 8 ! dial-peer voice 900 pots translation-profile outgoing international destination-pattern 900T port 0/0/0:15 ! dial-peer voice 852 voip translation-profile incoming from-gk incoming called-number 852. ! dial-peer voice 2300 voip destination-pattern [23]... session target ras tech-prefix 1# dtmf-relay h245-alphanumeric ! dial-peer voice 2301 pots translation-profile outgoing hq-sb preference 1 destination-pattern [23]... port 0/0/0:15 ! dial-peer voice 4220 voip destination-pattern 4220 session protocol sipv2 session target ipv4:142.102.66.253 incoming called-number 199[89].... dtmf-relay sip-notify codec g711ulaw

no vad ! presence presence call-list ! gateway ! sip-ua presence enable ! telephony-service sdspfarm units 2 sdspfarm transcode sessions 3 sdspfarm tag 1 sitec-xcode sdspfarm tag 2 sitec-cfb conference hardware no auto-reg-ephone authentication credential admin cisco max-ephones 5 max-dn 10 no-reg both ip source-address 142.1.66.254 port 2000 url services http://142.102.66.253/voiceview/common/login.do url authentication http://142.102.66.254/CCMCIP/authenticate.asp time-zone 42 voicemail 4220 max-conferences 12 gain -6 web admin system name admin password cisco dn-webedit transfer-system full-consult directory entry 1 4001 name Site C Phone 1 directory entry 2 4002 name Site C Phone 2 create cnf-files version-stamp 7960 Nov 27 2010 00:13:36 ! ephone-template 1 conference drop-mode local softkeys idle Redial Newcall Cfwdall Pickup Dnd softkeys seized Pickup Cfwdall Endcall Redial Meetme softkeys connected Hold Endcall Trnsfer Confrn Park ! ephone-dn 1 octo-line number 4001 no-reg primary description +85224044001 name sitec ph1 allow watch call-forward busy 4220 call-forward noan 4220 timeout 10 ! !

ephone-dn 2 octo-line number 4002 no-reg primary description +85224044002 name sitec ph2 allow watch call-forward busy 4220 call-forward noan 4220 timeout 10 ! ephone-dn 3 number A5001 no-reg primary intercom A5002 label "SC PH2" ! ephone-dn 4 number A5002 no-reg primary intercom A5001 barge-in label "SC PH1" ! ! ephone-dn 5 octo-line number 4321 no-reg primary conference meetme ! ephone-dn 6 number 1998.... no-reg primary mwi on ! ephone-dn 7 number 1999.... no-reg primary mwi off ! ephone 1 device-security-mode none mac-address 001B.D4A0.9428 ephone-template 1 username "scph1" password 12345 presence call-list type 7961 button 1:1 2:3 ! ephone 2 device-security-mode none mac-address 001B.D4C6.D8B6 ephone-template 1 username "scph2" presence call-list type 7961 button 1:2 2:4 3m1 ! ntp server 142.1.64.254