CCIE Voice Boot Camp Study Guide v3 0

Truly Unified
CCIE VOICE
Study Guide
v3.0
VoiceBootcamp
1
Overview
CCIE Voice Lab Overview
• A 8-hour, hands-on lab exam.

• 100-point lab exam. One must score 80 or above to pass.
• Candidate builds a voice network to supplied specifications
on a provided Voice equipment rack.
• UCM 7.0, Unity Connection, Presence, UCCX
• Physical cabling is done.
• IP routing protocol such as OSPF and Frame Relay is
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preconfigured.
This intense 5 day course is designed to prepare CCIE Voice candidates to successfully pass their CCIE
Voice practical lab examination. Over the duration of the course, candidates will be augmenting their
existing IP Telephony knowledge, remedy their problem areas and weaknesses, as well as, gain vital
test-taking strategies. This class is designed for candidates who are within 1 week to 9 months of their
CCIE Voice Lab date. The class does not cover introductory material and candidates are expected to
have minimum production knowledge of the topics covered in order to receive the full benefit of the class.
We strongly recommend students to have completed a majority of the labs in our CCIE Voice Workbook
prior to attending this course.
Course agenda
Agenda
Day 1
Section 01 Infrastructure
Section 02 Unified Communication Manager 7 Implementation
Section 03 Basic Unified Communication Express 7.0
Section 04 Voice Gateway - H323/SIP/MGCP/SIP Trunking/IP to IP Gateway/GK
Day 2
Section 05 Dial Plan - Call Routing/Hunt Group/CTI RP/Transformation Mask
Section 06 Dial Plan Feature - Intercom, Call park, Directed Call park, SIP Dial Rule
Section 07 Media Resources - Moh, Conference, Transcoding, Mobile Voice access, ANN
Section 08 SRST with CallManager Express, AAR, CAC/RSVP
Day 3
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Section 09 Integration with Unity Connection 7.0, Advanced Unity Connection Configuration
Section 10 Integrating with Unity Express 7.0
Section 11 integrating with Unified Contact Center Express/ Advanced Scripting
Day 4
Section 12 Integration with Cisco Unified Presence, Advanced Unified Presence & Microsoft OCS integration
Section 13 UC Application - IPMA, EM , Mobility, Single Number Reach, Mobile Access
Section 14 QoS
Day 5 8 hours Lab simulation
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Each candidate decides how they will study. Some have a goal to finish CCIE VOICE in 3 months while
others 3 years. Depending on your time schedule, you need to create a study plan. What you want to
cover on each steps.
Network Topology
CCIE VOICE
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• CCIE VOICE diagram

• Information Sheet containing DN, IP Address etc
Network Topology
Voice Lab Sample Topology
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For more updated Network Diagram please visit http://support.voicebootcamp.com

Chapter 1
Infrastructure and
Services
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Module Outline
• VLANS and VTP Server
• Configuring Cisco 6509 Catalyst Switches
• Configuring Cisco 35XX Catalyst Switches
• Configuring DHCP Servers.
• Configuring DHCP Relay Agent.
VLAN
VLAN
Si
Si
Distribution Edge IP Phone Tag

Layer Switch packet with 802.1q
for all voice traffic.
Data traffic remain
untag
PC VLAN = 500
Phone VLAN = 101
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trunk if it
is XL based
Switch
IP Phone
135.XX.66.0
Desktop PC:
135.XX.166.0
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• Virtual LAN. Group of devices on one or more LANs that are configured (using management software)
so that they can communicate as if they were attached to the same wire, when in fact they are located
on a number of different LAN segments
• 802.1Q Set of IEEE standards for the definition of LAN protocols.
• VTP : VLAN Trunking Protocol (VTP) is a Layer 2 messaging protocol that manages the addition,
deletion, and renaming of VLANs on a network-wide basis.
•Domain – Defines a management domain
•Password – Protect VTP communication
•Mode – define VTP mode Server, Client, Transparent
•V2 – enable or disable for Version 2.
• Must be configured first before assign them.
• Single Port can carry multiple VLAN if port is configured as a trunk port
• When IP phone is connected to an XL based switch all IP phone port must be Trunk and its native
VLAN must be set properly.
• VLANs do not allow any communication between them at Layer 2 unless InterVLAN routing is
configured to route traffic at Layer 3.
Step by Step Instructions for
VLAN
• Step 1 – CDP
• Step 2 - Create VLAN and VTP
• Step 3 – Assign Data VLAN to all IP Phone ports
• Step 4 – Assign Voice VLAN to all IP phone ports
• Step 5 – Router port must be trunk
• Step 6 - All voice port must be trunk if the switch is EtherSwitch
• Step 6 - All Trunk port must have native vlan set to data vlan
• Step 7 – Define DHCP Server to assign IP address
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• Some switches, by assigning VLAN to interfaces will create the VLAN in the VLAN databases
• Most new IOS requires you to create VLAN from configuration mode instead of old VLAN Databases.
Although VLAN database command may be available but try using configuration mode instead.
• If Switches are connected to another switch ensure that VTP is configured properly.
• NATIVE VLAN is mostly use for sending/receiving management information. NATIVE Vlan must be
configured properly in the switches as well as in router if router on the stick is in used.
• When IP Phone is connected to a
Cisco Discovery Protocol
CDP
Cisco Devices use CDP protocol to discover all

devices are connected to its port.
Cisco 3550 or XL Switch
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• Cisco Devices use CDP protocol to discover all devices are connected to its port.
• Layer 2 Protocol
• Cisco propriety protocol
• Identify by directly connected devices
• Used to identify name, ip address, which port connected to what etc
Data and Voice VLAN in
Catalyst 3500XL
2 Cisco Catalyst 3500XL

PC VLAN = 500
Voice VLAN = 101
Catalyst
3500XL IP Phone Desktop PC
Create VLAN 135.11.65.15 135.11.165.50
Switch# vlan data
If it is a EtherSwitch and/or XL
Switch(vlan) vlan 101 name RxVOICE
Switch, IP Phone port must be
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Switch(vlan) vlan 500 name RxDATA
TRUNK and NATIVE vlan must
Switch(vlan) vtp domain RACKXX
be set to data vlan
Switch(vlan) vtp server
Interface range FastEthernet0/1 - 4
switchport mode trunk
switchport trunk encapsulation dot1q
Assign VLAN to
switchport trunk native vlan 500
Port
switchport voice vlan 101
spanning-tree portfast
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Data and Voice VLAN in Cisco Catalyst 3500XL
• When configuring VLANS for Cisco IP phone connected to an XL based switch such as Cisco
3524XL or EtherSwitch NM module, IP phone ports must be trunk with 802.1Q trunking.
• Ensure that native VLAN is correctly set.
• Port where Router port is connected must be configured to trunk multiple VLAN and ensure
NATIVE vlan is configured properly.
• Ensure VTP is also configured properly if required
NOTE:
Spanning Tree on Trunk port has no effect. Therefore if you are ask to define port fast then do not
trunk the port. It is assume that when asked for portfast, Switch will not be an XL or EtherSwitch
module
Data and VOICE VLAN
Data and Voice VLAN in

Catalyst 3550 L3
3 Catalyst 3550 L3 Switch

Voice VLAN = 101 PC VLAN = 500
Catalyst
3550
IP Phone Desktop PC
Create VLAN 135.11.65.15 135.11.165.50
802.1Q
Switch# vlan dataTrunk
Make sure ROUTER PORT
Switch(vlan) vlan 101 name RxVOICE
Switch(vlan) vlan 500 name RxDATA Is trunk port with native vlan
Switch(vlan) vtp domain RACKXX set to data vlan
Switch(vlan) vtp server
interface FastEthernet2/0
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no ip address
Assign VLAN to switchport access vlan 500
Port switchport voice vlan 101
spanning-tree portfast
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Data and VOICE VLAN – Catalyst 3550 L3 Switch
• IP phone connected to Cisco 3550 SMI or EMI does not require to trunk IP phone ports. Simply
assign Access and Voice VLAN
• Router port must be trunk if inter-vlan routing is not being used.
Network Services – NTP, DHCP, DNS
Networks Services
• DNS configuration is required if name resolution is required

• Network Time Protocol server must be configured.
• DHCP used to automate network access
• MS DHCP or IOS DHCP
DHCP server needs to provide the following:
IP Address and network mask
Default Gateway
Option 150, TFTP server IP address
DNS Server (optional)
Use ip helper-address to forward DHCP request to DHCP Server
Can be locally implemented on IOS router just incase WAN failure occurs.
• CDP is required in order for IP Phone to communicate with AVVID
network
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DNS server
• DNS enables the mapping of host names and network services to IP addresses within a network
or networks.
• DNS server(s) deployed within a network provide a database that maps network services to
hostnames and, in turn, hostnames to IP addresses.
• Devices on the network can query the DNS server and receive IP addresses for other devices in
the network, thereby facilitating communication between network devices.
• Complete reliance on a single network service such as DNS can introduce an element of risk
when a critical IP Communications system is deployed.
• If the DNS server becomes unavailable and a network device is relying on that server to provide
a hostname-to-IP-address mapping, communication can and will fail. For this reason, It is highly
recommends that you do not rely on DNS name resolution for any communications between
Cisco Unified CallManager and the IP Communications endpoints.
• DHCP provides the following information to end devices
•IP Address
•Subnet Mask
•Option 150 IP address for TFTP
•Default Gateway for device to access other networks.
• IP Helper address is require for centralized DHCP deployment or when IP devices and DHCP
server are on two different subnet.
• Multiple option 150 can be assign to IP devices. To configure multiple Option 150
•MS DHCP – use Array when creating Option 150
•IOS – define two or more IP address one after another.
• CDP must be enable in order for IP phone to work properly in Cisco environment.
UCM 7.0 DHCP
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Dynamic Host Configuration Protocol
Dynamic Host Configuration Protocol (DHCP) server enables Cisco Unified IP Phones, connected to
either the customer's data or voice Ethernet network, to dynamically obtain their IP addresses and
configuration information
Procedure
• From Cisco Unified Serviceability, choose Tools > Service Activation.
• The Service Activation window displays.
• Choose the Cisco Unified Communications Manager server from the Servers drop-down list box and
click Go.
• Choose Cisco DHCP Monitor Service from the Unified CM Services list and click Save.
Note : If the service is already activated, the Activation Status will display as Activated.
• The service gets activated, and the Activation Status column displays the status as Activated
DHCP Servers
Server where DHCP will be hosted
DNS Server
TFTP Server
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Procedure
• Choose System > DHCP > DHCP Server

• Perform one of the following tasks:
• To add a DHCP server, click Add New.
• To update a server, find the server by using the procedure in the Finding a DHCP Server topic.
• To copy a server, find the server by using the procedure in the Finding a DHCP Server topic, select the
DHCP server that you want by checking the check box next to the server name, and click the Copy
icon.
• The DHCP Server Configuration window displays.
• Click the Save icon that displays in the tool bar in the upper, left corner of the window (or click the Save
button that displays at the bottom of the window) to save the data and to add the server to the
database.
DHCP Subnet
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Procedure
Choose System > DHCP > DHCP Subnet.
The Find and List DHCP Subnets window displays.
To find all records in the database, ensure the dialog box is empty; go to Step 3.
To filter or search records:
From the first drop-down list box, select a search parameter.
From the second drop-down list box, select a search pattern.
Specify the appropriate search text, if applicable.
• Note : To add additional search criteria, click the + button. When you add criteria, the system searches
for a record that matches all criteria that you specify. To remove criteria, click the - button to remove the
last added criteria or click the Clear Filter button to remove all added search criteria.
• Click Find.
• All or matching records display. You can change the number of items that display on each page by
choosing a different value from the Rows per Page drop-down list box.
• Note : You can delete multiple records from the database by checking the check boxes next to the
appropriate record and clicking Delete Selected. You can delete all configurable records for this
selection by clicking Select All and then clicking Delete Selected. From the list of records that display,
click the link for the record that you want to view.
• Note : To reverse the sort order, click the up or down arrow, if available, in the list header. The window
displays the item that you choose.
Networks Services: DHCP
DHCP Server IP phone request for IP

(135.7.100.20) Via DHCP Broadcast
PSTN
CallManager
SFO
IP WAN
Toronto
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Interface vlan 101
ip address 135.7.65.240
ip helper-address 135.xx.100.11
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DCHP Server
• DHCP is used by hosts on the network to obtain initial configuration information, including IP
address, subnet mask, default gateway, and TFTP server address.
• DHCP eases the administrative burden of manually configuring each host with an IP address and
other configuration information.
• DHCP also provides automatic reconfiguration of network configuration when devices are moved
between subnets.
• Use IP enabled devices to use DHCP whenever possible to ease administration.
• DHCP server should be redundant so incase of failure alternative DHCP server is available to
provide IP addresses.
• DHCP Scope must provide necessary address information such as
•IP Address of the end devices
•Subnet mask
•Default Router (gateway)
•TFTP IP address via Option 150
• Cisco IP phone is capable of having maximum of two TFTP addresses.
• Router may block DHCP traffic due to broadcast if end devices and DHCP servers are not in the
same subnet therefore use of IP HELPER-ADDRESS under inbound interface of each router is
required in order to relay DHCP traffic back to the DHCP Server.
Networks Services: IOS DHCP
Most IOS Router can act as a DHCP Server Exclude IP address
ip dhcp excluded-address 135.XX.67.1 135.XX.67.14

ip dhcp excluded-address 135.XX.67.51 135.xx.67.254
ip dhcp pool VOICE VLAN 10X VLAN 500

network 135.XX.67.0 255.255.255.0
default-router 135.XX.67.240
option 150 ip 135.xx.67.240
!
!
interface fastEthernet0/0.10X (where X is Rack) VLAN 10X
VLAN 500
Enacapsulation dot1q 10X
ip address 135.xx.67.240 255.255.255.0
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! UK Office
interface fastEthernet0/0.500
Encapsulation dot1q 500 native vlan
ip address 135.XX.167.240 255.255.255.0
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IOS DHCP Server
• Cisco router has the capability of becoming DHCP server

• Ensure you exclude the IP address first before creating the DHCP scopes
• IP helper-address may be require to configure relay if end device and dhcp server are not in the
same subnet
NTP Configurations
Toronto – Eastern Time Zone

r7tor(config)#clock timezone EST -5
r7tor(config)# ntp server
135.11.11.11
SFO – Pacific Time Zone

r7sfo(config)#clock timezone PST -8
r7sfo(config)#ntp server 135.11.11.11
UK – GMT 0
r7uk(config)#clock timezone GMT 0
r7sfo(config)#ntp server 135.11.11.11
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NTP configurations
NTP is often required by many network devices to provide a synchronized time

UCM NTP Server
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Unified Communication
Manager 7.0
CallManager Basic
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• Cisco CallManager serves as the software-based call-processing component of the Cisco IP Telephony
Solutions for the Enterprise
• The Cisco CallManager system extends enterprise telephony features and functions to packet
telephony network devices such as IP phones, media processing devices, voice-over-IP (VoIP)
gateways, and multimedia applications. Additional data, voice, and video services such as unified
messaging, multimedia conferencing, collaborative contact centers, and interactive multimedia
response systems interact through Cisco CallManager open telephony application programming
interface (API).
Deployment Models
Centralized Call Processing
AVVID Application
Server
SRST
PSTN
CallManager SFO
Cluster
IP
backbone
CME
Toronto
UK
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CME Router
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In the Multisite WAN Design, centralized call processing consists of a single call processing system
That provides services for many sites and uses the WAN or dedicatred leased line to transport IP
telephony traffic between the sites. The IP WAN also carries call control signaling between the
central site and the remote sites.
Benefits
•Call Processing take places in head office
•All signalling cross IP WAN even for calls between two IP Phone in branch offices
•CallManager can provide centralized or distributed DSP resources. I.E
•Headoffice can provide Conference Services from HQ DSP as primary and use DSP resources
in branch office router as a backup.
•Local resources can use local DSP resources such as all Branch office IP phone can use DSP
resources from the local router as oppose to getting the resources from CallMananagers.
Simple Call Process
Unity 4.x
Voice Mail
Exchange 2K
1 Call Setup CCM
SFO 4 IPWAN
Ring Back
2 E.164 Lookup
Phone 1
Call
Connect Setup
3
6 RTP 4
Stream Ring
PSTN
5 Off Hook
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TOR
Phone 2
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• Phone 1 dials Phone 2

• Callmanager does a E.164 lookup and find that phone 2 is a registered device.
• CallManager will initiate Call setup to Phone 2
• CallManager will then send a ring to Phone 2 and ring back to Phone 1
• Phone 2 picks up the phone and goes to off hook
• RTP streem is between the IP Phones
NOTE:
• While IP phone has established RTP stream with another IP phone, if Callmanager goes down, IP
phone will remain up and user will be able to continue to talk.
• If IP phone is behind NAT or Firewall, one way audio can occur if one side is blocking traffic from other
side. Ensure RTP is passes through the NAT and Firewall.
CallManager Cluster &
Redundancy
• CallManager Group defines redundancy.
• Group can have up to 3 CCM Server.
• First server in the list is the Active CCM
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VC
Publisher
Cluster Standby CCM
ccmpub
CCM
Group
Default Primary CCM ccmsub
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CallManager Cluster and Redundancy
• A Cisco CallManager group specifies a prioritized list of up to three Cisco CallManagers. The first
Cisco
• CallManager in the list serves as the primary Cisco CallManager for that group, and the other
members of the group serve as secondary and tertiary (backup) Cisco CallManagers.
• Each device pool has one Cisco CallManager group assigned to it.
• Device first attempts to connect to the primary (first) Cisco CallManager in the group that is
assigned to its device pool
• To support up to 7,500 phones you should have at least 2 servers. As you can see from the
figure above, one server will be the publisher and the secondary or backup Cisco CallManager.
• The second server will be a subscriber server and the primary Cisco CallManager to handle all
the call processing.
CCM Device Registration
Device with Extension ‘3001’ Is

TCP Connect Registered to Me (ccmpub)
(Active) UCMPUB
SCCP
KeepAlive/30s
3001 UCMSUB-A
UCMSUB-B
CCM GROUP A
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1: ccmpub
2: ccmsubA
3: ccmsubB
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• This is second type of intra-cluster communication.

• When a device registers to a Cisco CallManager cluster, the Cisco CallManager communicates
with all the other Cisco CallManager servers in the cluster as shown in the figure above. After the
device registers with the Cisco CallManager, it sends a TCP keep alive every 30 seconds and
sends a TCP connect to the secondary Cisco CallManager.
• The next figure shows what happens when a Cisco CallManager becomes unavailable.
CCM Device Registration
(cont’td)
UCMPUB Device with Extension ‘3001’

is UN-registered to Me (ccmpub)
X
SCCP KA
Device with Extension ‘3001’

3001 UCMSUB-A is UN-registered to Me (CCM D)
UCMSUB-B
Cisco CallManager List
1: UCMPUB
2: UCMSUB-A
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3: UCMSUB-B
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• When a Cisco CallManager fails, it will send a message to all Cisco CallManager servers in the cluster
making them aware that the devices registered to it, have un-registered. The secondary Cisco
CallManager accepts the registration from the device, then announces to all the Cisco CallManager
servers in the cluster that the device is now registered to it. The device then establishes a TCP keep
alive to the secondary Cisco CallManager and also a TCP connect to the tertiary.
• You can only define no more then 3 callmanager in a group. If a branch office loose connection to
Primary CallManager it will fall back to secondary or tertiary however if a branch office loose IP
connectivity to any CallManagers then Branch office can rely on SRST.
Tools Service Active
Select Service Activation
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Cisco Unified Serviceability service management includes working with feature and network services and
servlets, which are associated with the Tomcat Java Webserver. Feature services allow you to use
application features, such as Serviceability Reports Archive, while network services are required for your
system to function.
Procedure
• Choose Tools > Service Activation.
• The Service Activation window displays.
• From the Server drop-down list box, choose the server where you want to activate the service; then,
click Go.
• For the server that you chose, the window displays the service names and the activation status of the
services.
• To activate all services in the Service Activation window, check the Check All Services check box.
CallManager Server
DNS-Less Environment
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Enables Cisco IP Phones and other CCM-controlled devices to
contact the CCM without resolving a DNS name
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• Complete reliance on a single network service such as DNS can introduce an element of risk
• If the DNS server becomes unavailable and a network device is relying on that server to provide a
hostname-to-IP-address mapping, communication can and will fail.
• Cisco highly recommends that you do not rely on DNS name resolution for any communications
between Cisco Unified CallManager and the IP Communications endpoints.
Call Manager Configuration Example
Device Registration and Redundancy
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• Use Cisco CallManager configuration to specify the ports and other properties for each Cisco
CallManager that is installed in the same cluster.
• Use to define Auto-Registration
Define Group to provide Redundancy
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• Atleast one group must have Auto registration enable. This allow devices registering for the first time
to register to the CallManager. It is often suggested that default group should have Auto Registration
turn on. The reason behind this is when a device registering for the first time, it does not know which
group to join. Therefore default group should be used to auto-register.
• Once device has been auto-register then it can be moved to right device group.
• Group priority is based on TOP DOWN approach. Active CallManager or Primary CallManager is the
CallManager that is top of the list. Then Secondary or backup callmanager is the next one in the list.
Time/Date
•
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Group – Define a name for the Time zone such as Eastern or New York – EST etc.
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• Time zone – select a predefine timezone from the drop down list
• Separate – How you want to format the time for example: Jan – 1 – 5007
• Date format define how you want the date to be display month first following by day and year.
• Time format either in 12 hour regular format with AM/PM or military format where 6 PM is = 18
Region
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• Regions used to specify the bandwidth that is used for audio and video calls within a region and
between existing regions
• The audio codec determines the type of compression and the maximum amount of bandwidth that is
used per audio call.
• The video call bandwidth comprises the sum of the audio bandwidth and video bandwidth but does not
include overhead.
• Allows maximum of 500 region per Clusters

Device Pools
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Use device pools to define sets of common characteristics for devices. You can specify the
following device characteristics for a device pool:
•Cisco CallManager group
•Date/time group
•Region
•Softkey template
•SRST reference
•Calling search space for auto-registration
•Media resource group list
•Music On Hold (MOH) audio sources
•User and network locales
•Connection monitor duration timer for communication between SRST and Cisco
CallManager
•MLPP settings
Device Pools (cont’d)
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• Device Pool is like a common set of configurations applied to all the devices in a group.
• Each physical location should have a unique device pool
• Device Pool is be used by device mobility
• For a single site, you can disable SRST features for certain phone by using device pool.
• Every device in a certain physical location must be in its own device pool
Enterprise Parameters
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• Enterprise parameters provide default settings that apply to all devices and services in the same
cluster. (A cluster comprises a set of Cisco CallManagers that share the same database.) When you
install a new Cisco CallManager, it uses the enterprise parameters to set the initial values of its device
defaults such as URL that IP phone use to access services
• Often Enterprise parameters require some changes such as modifying URL so that IP phone can reach
the devices properly.
• You can also restrict what user can do to their phone if they have access to CCMUSER web pages.
• Many of the enterprise parameters rarely require change.
• Make sure you fully understand the parameter before you change any value unless you speak with an
TAC agent.
• DNS Less Environment where IP phone does not depend on DNS, you must ensure that all HTTP
reference must point to an IP address instead of a hostname or NetBIOS name.
• Enterprise parameter can also be used to define what option user has when they login to their IP phone
via web
Device Registration and Redundancy (cont’d)
Add New IP Phones
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Cisco IP Phones as full-featured telephones can plug directly into your IP network. You use the
Cisco CallManager Administration
• You can automatically add phones to the Cisco CallManager database by using auto-registration,
manually add phones by using the phone configuration windows
• To Add hundreds of IP phone together you can use CallManager Builk Administrative Tools
• CallManager use mac address of the device to register it in the database therefore you can move your
IP Phone to any IP network in the world as long as it has connection to CallManager, it will register and
get all the configurations.
Device Level Configuration
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• Manually Added phone require the MAC address of the IP Phone. CallManager use MAC Address
instead of IP address. Therefore IP Phone can be mobile.
• Device Pool must be define which basically inherit all the settings require for that IP Phone
• You must define a SoftKey Template which modifies the LCD screen
• Define a Phone Button Template to allow 1 or more lines.
• Once Phone has been added, you need to define a Directory Number which is the extension number of
this Phone.
IP Phone Line setting
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• Directory Number is the extension number of this IP phone.

• IP phone can be secured by defining a partition
• VoiceMail Profile allow you to select a specific Voice Mail profile or use the default. NONE means
default.
• Auto Answer allow this IP phone to answer call automatically when there is an inbound call to extension
3001
• Administrator has the ability to define a different music file to be played during Hold. User Hold Audio
source plays when one user put another user on hold. Network Hold audio source is played when call
is on hold due to Transfer, Call Park, Conference etc.
Unified Communication Express 7.0
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CCME: Cisco Call Manager Express
PSTN
IP WAN
• Call Manager in an Cisco IOS router with special IOS.

• Router provides call processing to Cisco IP phones.
• Same router also serves as an PSTN gateway: it terminates ip
packet voice to TDM voice and vice versa. It can also be used
as routing devices.
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• Cisco Unified CME is an excellent choice for a single-site, standalone office.
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• Leading-edge productivity features and improved customer service IP-based applications, such as XML
services, can also be deployed easily over this converged infrastructure.
• In other word, CME is a Call Manager in an Cisco IOS router.
• Router provides call processing to SCCP endpoints such IP phones.
• Same router also serves as an PSTN gateway: it terminates ip packet voice to TDM voice and vice
versa
CME Setup
Entering Telephony mode
telephony-service
load 7960-7940 P00308000500 Define Phone loads for
max-ephones 100 max-dn 240 upgrade/downgrade
ip source-address 135.Y.67.240 port 5000 Define max number of phone
ip qos dscp ef media
Define what IP to bind CME to
ip qos dscp cs3 signal
create cnf QoS Settings for voice traffic
Create the configuration files
cnf-file location flash:

cnf-file perphone
auto-reg-ephone
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• Load command defines what firmware to load for particular type of phone
• Max-ephone define how many maximum number of phone to register. Now if you reduce max-ephone
compare to what is registered, all existing phone will not be disconnected right away. They will
continue as normal until they reboot or reregister
• IP source-address defines what IP address you want the Callmanager Express to bind to.
Extra configurations
To define a location other than system:/its for storing configuration files for per-phone and per-phone type
configuration files, perform the following steps.
cnf-file location flash:  This tells the CME to store all the configs in the flash
cnf-file perphone or perphonetype  This tells the CME to configuration file will be per
phone basis or type
auto-reg-ephone - Can be used to prevent SCCP phone from registering automatically

UCME – Redundant Router
PSTN
telephony-service
ip source-address 135.7.67.240 port 5000 secondary
135.Y.67.241
IP WAN
voice-port 3/0/0
signal ground-start
incoming alerting ring-only
Backup CME router telephony-service (2nd router)

Must have Voice ip source-address 135.7.67.240 port 5000 secondary
port Ring number 135.Y.67.241
set To higher
then primary voice-port 3/0/0
VoiceBootcamp
signal ground-start
incoming alerting ring-only
ring number 3 41
A second Cisco Unified CME router can be configured to provide call-control services if the primary Cisco
Unified CME router fails. The secondary Cisco Unified CME router provides uninterrupted services until
the primary router becomes operational again
When a phone registers to the primary router, it receives a configuration file from the primary router.
Along with other information, the configuration file contains the IP addresses of the primary and
Secondary Cisco Unified CME routers. The phone uses these addresses to initiate a keepalive (KA)
Message to each router. The phone sends a KA message after every KA interval (30 seconds by default)
To the router with which it is registered and after every two KA intervals (60 seconds by default) to the
Other router. The KA interval can be adjusted
Ring number
Required only for the secondary router) Sets the maximum number of rings to be detected before
answering an incoming call over an FXO voice port. • Number—Number of rings detected before
answering the call. Range is 1 to 10. Default is 1. Note For an incoming FXO voice port on a secondary
Cisco Unified CME router, set this value higher than is set on the primary router. We recommend setting
this value to 3 on the secondary router.
SIP: Setting Up Cisco Unified CME
Configure terminal
Voice register pool  This command difine CME to support SIP

mode cme
source-address ip-address 135.Y.67.240
tftp-path http://www.voicebootcamp.com/files
max-pool 25
authenticate all realm voicebootcamp.com
voice register dn 2 Define an extension

number 6001
call-forward b2bua busy 6600
huntstop channel 3
! Assign the extension
voice register pool 123 to a Phone
busy-trigger-per-button 2
id mac Y.Y.Y.Y
type 7961
number 1 dn 2
VoiceBootcamp 42
If your Cisco Unified CME system supports SCCP and SIP phones, do not connect your SIP phones to
your network until after you have verified the configuration profile for the SIP phone
Configuration Guide
Voice register pool
mode cme  This command define CME to support SIP
source-address ip-address 135.Y.67.240 this is the IP where CME will listen for IP Phone to register
Tftp-path  This is where CME download the phone configuration from for the IP Phone. Example: tftp-
path http://www.voicebootcamp.com/files
Max-pool  defines how many phone that can be registered. (just like max-ephone)
SCCP – Setting UP CME for SCCP
telephony-service
ephone-dn 2
max-ephones 100 max-dn 240 number 6001
ip source-address 135.Y.67.240 port 5000
ephone 1
Mac-
button 1:2
ephone-dn 3 dual-line
Dual Line number 6002
Octo-line
ephone-dn 4 octo-line
6001 6002 number 6003
6003
ephone 2
MAC address: Mac-address Y.Y.Y.y
Y.Y.Y.Y button 1:3 2:4
VoiceBootcamp 43
CME IP Phone settings
• Phones in Cisco Unified CME

• Directory Numbers
• Monitor Mode for Shared Lines
• Watch Mode for Phones
• PSTN FXO Trunk Lines
• Codecs for Cisco Unified CME Phones
• Analog Phones
• Remote Teleworker Phones
• Busy Trigger and Channel Huntstop for SIP Phones
• Digit Collection on SIP Phones
• Session Transport Protocol for SIP Phones
VoiceBootcamp
• Ephone-Type Configuration
44
Phone & Directory Number
Ethernet Phone or voice-register pool Single Line
•Used by a Phone it self ephone-dn 2
•Each phone must have a ephone X configure number 6001
Directory Number ephone 1

• Number assign to line Mac-
button on the phone button 1:2
• Single Line
• Dual Line
• Octo-Line
Dual Line number 6002
Octo-line
ephone-dn 4 octo-line
6001 6002 number 6003
6003
ephone 2
MAC address: Mac-address Y.Y.Y.y
MAC address: X.X.X.X
VoiceBootcamp
Y.Y.Y.Y button 1:3 2:4
45
An ephone is an Ethernet phone, and an ephone-dn is an Ethernet phone directory number. In CM

Express, an ephone is a logical configuration and settings for a physical phone, and the ephone-dn is a
destination number that can be assigned to multiple ephones.
An ephone-dn always has a primary directory number, and it may have a secondary one as well. When
you create an ephone-dn, you can specify it as single line (the default) or dual line. A single line can
terminate one call; a dual line can terminate two calls at the same time. This is necessary for call waiting,
consultative transfer, and conferencing features to work. When you create an ephone-dn, the router
automatically creates POTS dial peers to match
NOTE:
• There is a maximum number of ephone-dns that a given platform will support; this is controlled by the
hardware capacity and by licensing.
• The max-dn <max-dn-value> command must be set to create ephone-dns – default zero
• Once max-dn is define router will automatically reserve enough memory to support it regardless
if they are being used or not.
Ephone
• An ephone is the logical configuration of a physical phone
• Each ephone is given a tag to uniquely identify it. (like a sequence number 1, 2, 3 and 4…)
• Each ephone is given a tag to uniquely identify it. The MAC address of the phone ties it to the ephone
configuration (in each ephone you define the mac address of an particular IP Phone. That’s how a
physical IP Phone is associated with a ephone)
• All the IP Phone model type are automatically detected (if augo register is enable) except 7914
• Each different model of IP Phone has a different number of buttons (the top button is always numbered
" 1 , “)
Example:
• r o u t e r ( c o n f i g ) # ephone 2
• r o u t e r ( c o n f i g - e p h o n e ) # mac-address XXXX.YYYY.AAAA
• r o u t e r ( c o n f i g - e p h o n e ) # type 7960 addon 1 7914
• r o u t e r ( c o n f i g - e p h o n e ) # button 1:2
Directory Number (extension)

• Directory Number
• Extension number assigned to IP Phone
• ephone-dn is configured to assign extension to phone
• Each ephone-dn can be
• Single Line – 1 calls per line
• Dual Line – 2 calls per line (call waiting)
•If line is shared among two Phone, phone that answer the call will take control of both
channel
VoiceBootcamp
• Octo-line – 8 calls per line
• if DN is shared among multiple phone, only one channel is seized by the phone that
answer the call
•Other user will see Remote-In-Use on shared line
Line Comparison
VoiceBootcamp 46
Single line:
• This ephone-dn creates a single virtual port. Although you can specify a secondary number, the phone
can terminate only one call at a time, so it cannot support call waiting. It should be used when there is
one phone button for each PSTN line that comes into the system. It is useful for things like paging,
intercom, call-park slots, MoH feeds, and MWI.
r1uk(config)#ephone-dn 1
r1uk(config-ephone-dn)#number 6001
There can only be one call at the above number 6001. If there is another incoming call while line
is already connected user will hear a fast busy. Call waiting in this scenario is disable
Dual line:
• The dual-line ephone-dn can support two call terminations at the same time and can have a primary
and a secondary number. It should be used when a single button supports call features like call waiting,
transfer, and conferencing. It should not be used for lines dedicated to intercom, paging, MoH feeds,
MWI, or call park. It can be used in combination with single-line ephone-dns on the same phone.
r1uk(config)#ephone-dn 10 d u a l - l i ne
r1uk(config-ephone-dn)#number 6002
Extension 6002 can now handle two call simultanously. Therefore call waiting is now enable.
Dual number:
• This ephone-dn has a primary and secondary number, making it possible to dial two separate numbers
to reach the phone. It can be either a single- or dual-line ephone-dn; it should be used when you want
to have two numbers for the same button without using more than one ephone-dn.
r1uk(config)#ephone-dn 10 dual-line
r1uk(config-ephone-dn)#number 6002 secondary 6003
If some one dials 6002 or 6003, it will ring the same line ephone-dn 10
Shared ephone-dn:
• The same ephone-dn and number appears on two separate phones as a shared line, meaning
thateither phone can use the line, but once in use the other cannot then make calls on that line. The line
will ring on all phones that share the ephone-dn, but only one can pick up. If the call is placed on hold,
any one of the other phones sharing the line can pick it up.
Overlay ephone-dn:
• An overlay consists of two or more ephone-dns (up to 25) applied to the same button; all these ephone-
dns must be either single or dual line
VoiceBootcamp
SIP None Shared-Line
(Nonexclusive)
• SIP DN can also be shared line

• CME must be configured for SIP based network
voice register dn 2
number 6001
huntstop channel 3
!
voice register pool 123
id mac Y.Y.Y.Y
type 7961 6001
number 1 dn 2
VoiceBootcamp
MAC address:
Y.Y.Y.Y
47
• SIP based DN can be shared among multiple phone

• All phones sharing the directory number can initiate and receive calls at the same time
• After a phone answers a call, the ringing stops on all phones and the call-waiting tone plays for
other incoming calls to the connected phone
• Any shared-line phone user can resume the held call
• If the call is placed on hold as part of a conference or call transfer operation, the resume is not
allowed.
• Shared lines support up to 16 calls
SIP Shared Line
busy-trigger-per-button
voice register dn 2
number 6001
shared-line max-calls 6
huntstop channel 6
!
id mac Y.Y.Y.Y
type 7961 Phone 1 Phone 2
number 1 dn 2
6001 6001
!
busy-trigger-per-button 3 MAC address:
id mac X.X.X.X X.X.X.X MAC address:
type 7965 Y.Y.Y.Y
number 1 dn 2
VoiceBootcamp 48
• In this scenario first two calls will arrive on Phone 1 and 3rd call will arrive on Phone 2 because of busy-
trigger-per-button configuration
Watch Mode for Phones
• Provide BLF (Busy Lamp Field) notification for all the

lines on another phone. E.G. Assistant has a speed dial
with BLF setup of the manager phone. Assistance can
have a visual notification of manager’s line status
• Line that are set for watched mode can not be used to
make and receive calls
• Incoming calls on a line button that is in watch mode do
not ring and do not display caller ID or call-waiting caller
ID
VoiceBootcamp 49
Presence is defined using BLF feature of CME.

CME - Shared Lines
• “In shared line” call distribution to ring multiple phones at

same time
• Same ephone-dn entry is assigned to multiple phones
• Each ephone-dn can only handle one call at a time. Once
the ephone-dn is in use, no further calls are accepted on the
ephone-dn.
VoiceBootcamp 50
SCCP Shared Line
Inbound call to 6011
09:00 06/500/05 6001 09:00 06/500/05

6002
6001
6012
6011
UK phone 1 UK phone 2 6011
Cisco CME
Cisco CME
ephone-dn 10
number 6011  Shared DN
ephone 1 ephone 2
mac-address 2222.2222.2223 mac-address 2222.2222.2222
VoiceBootcamp
button 3:10 button 3:10
51
• Ephone-dn 5 is assigned to line 2 of both phone 1 and phone 2

• Incoming calls to DN 5 will ring both IP phone at once
• If Phone 1 answer the call, Phone 2 can not use the 2nd line to make calls
Sequential Different DNs using
Call Forward
• Forwards call to another DN if the intended DN does not

answer or is busy
• Can be another DN on the same phone or on a different
phone
• One phone or DN rings at a time.
• Key Commands:
–Call-Forward Busy
–Call-Forward noan
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52
• Using Call-Forwared Busy and No Answer, an incoming call be redirected to another extension
on the same phone or a different phone or to a voicemail number.
• Call-Forward Busy is used when line is in use

• Call-Forward noan is used when line is not answering the call. In this case a timer to required to
decide after how long before the system will configure a line busy.
CallManager Express Call Distribution/Hunting:
Sequential Different DNs using Call Forward
Inbound call to 6001 ephone-dn 11

number 6001
09:00 06/10/07 6001 call-forward busy 6002
6001 Call-forward noan 6002 timeout 18
ephone 1
IP phone 1
VoiceBootcamp Inc button 1:1
If phone 1 is
busy or no Advice callmanager
answer, call is express to forward calls
forwarded to to 6002 if 6001 is busy or ephone-dn 2
6002 in this does not answer after 18
case Phone 2 number 6002
seconds.
call-forward busy 6003
Call-forward noan 6003 timeout 18
09:00 06/10/07 6002
ephone 2
6002
button 1:2
IP phone 2
VoiceBootcamp
VoiceBootcamp Inc.
53
• In Sequential Different DN call comes to an extension such as 6001 and if it is busy and/or does
not answer within 18 seconds, call will get forwarded to the next extension.
• Notice how call forward is based on an extension number but not the DN number.
• You can forward call using call-forward command to either a voice mail pilot number, to a number
that is in CallManager or even to a PSTN number using properly prefixes
CallManager Express
Call Distribution: Sequential Same DN
• Create multiple ephone-dn entries with the same DN

number and assign to different phones
• Control Sequential hunt order using
preference
[no] huntstop
huntstop channel
• Only one phone rings at a time
VoiceBootcamp 54
• Preference – 0 is the higest 10 is the lowest. Decide one gets first priority.
• Huntstop – Prevent system from continue to search for a matching pattern. When a ephone has
a no hunstop configured, basically when that phone is busy, CME will instruct the system to
continue to search for ephone with the same number.
• Each dual-line ephone-dn has 2 channel per line such as for call waiting. Huntstop channel
means stop the 2nd channel from receiving calls.
CallManager Express Call Distribution:
Sequential Same DN
Preference 0 is the
Inbound call to 6001 highest priority and
ephone-dn 1
the default value, it
number 6001 does not appear in
09:00 06/500/05 configuration
6001
no huntstop
6001 preference 0
IP phone 1 ephone 1
VoiceBootcamp Inc. mac-address 3001.3001.3001
If 6001 on If DN is not available and button 1:1
phone 1 is there is a match and no hunt-
busy, ring stop configure the call will go
next to the next DN based on
match preference. For this work, ephone-dn 2
both DN must have the same
number. number 6001
09:00 06/500/05 preference 1
6001
ephone 2
6001
mac-address 2222.2222.2222
IP phone 2
VoiceBootcamp
button 1:2
VoiceBootcamp Inc.
55
• When two or more DN has the same number assign to multiple IP hone, you can route calls
using hunt stop and preference command.
• Huntstop prevents an incoming call from rolling over to another ephone-dn if the called ephone-
dn is busy or does not answer. Use of no huntstop allow to rolling over to another ephone-dn
CallManager Express Dual-line
Huntstop Channel
• Channel huntstop works in a similar way for the

two channels of a dual-line ephone-dn
• Allow you to disables call-waiting on a dual-line
DN
• Reserves the second channel of a line for outgoing
calls such as transfer and conference
VoiceBootcamp
56
• Channel huntstop works in a similar way for the two channels of a dual-line ephone-dn. If it is
enabled, channel huntstop keeps incoming calls from hunting to the second channel if the first
channel is busy or does not answer.
• This keeps the second channel free for call transfer, call waiting, or three-way conferencing.
• Channel huntstop also prevents situations in which a call can ring for 30 seconds on the first
channel of a line with no person available to answer and then ring for another 30 seconds on the
second channel before rolling over to another line.
CCME Dual-line with Huntstop
Channel
09:00 06/500/05
6001 Incoming Call to 6001
6001
IP phone 1 6001 Line 1 6001
VoiceBootcamp Inc. Channel #1
number 6001 Channel #2
no huntstop
huntstop channel
number 6001 Line 2 6001
huntstop channel
Channel #1
preference 1
ephone 1 Channel #2
mac-address 5001.5001.5001
VoiceBootcamp
button 1:1 4:6
57
• Prevents incoming calls from hunting into the second channel of a dual-line DN
• Allow you to disables call-waiting on a dual-line DN
• Reserves the second channel of a line for outgoing calls such as transfer and conference
CCME Dual-line without Huntstop
Channel
09:00 06/10/07 6001
Incoming Call to 6001
6001
6001
UK phone 1 Line 1 6001
6001
VoiceBootcamp Inc.
Channel #1
Channel #2
number 6001
no huntstop
number 6001
Line 2 6001
preference 1
ephone 1 Channel #1
mac-address 3001.3001.3001
button 1:1 2:6 Channel #2
VoiceBootcamp 58
• Without huntstop channel, 2nd call will arrive in Channel # 2 in Line 1 while 3rd call will go to Line
2 channel # 1
• This means Call Waiting is enable.
CCME ephone-hunt
ephone-hunt allows CCME administrators to:

• Define a pilot number for a hunt group
• Sequential mode: specifies an ordered list of extension
numbers to sequentially hunt through
• Peer mode: specifies a random start point in a circular
list of extension numbers
• Longest Idle: specifies who is idle for long.
• Define a final destination to forward the call to if the call
is not answered or all members are busy
VoiceBootcamp 59
• There are three different kinds of ephone hunt groups.
•Sequential ephone hunt groups—Ephone-dns always ring in the left-to-right order in

which they are tried when the pilot number is called. Maximum number of hops is not a
configurable parameter for sequential ephone hunt groups.
•Peer ephone hunt groups—The first ephone-dn to ring is the number to the right of
the ephone-dn that was the last to ring when the pilot number was last called. Ringing
proceeds in a circular manner, left to right, for the number of hops specified when the
ephone hunt group was defined.
•Longest-idle ephone hunt group—Calls go first to the ephone-dn that has been idle
the longest for the number of hops specified when the ephone hunt group was defined.
The longest-idle is determined from the last time that a phone registered, reregistered,
or went on-hook.
Ephone hunt
r5uk(config)#ephone-hunt 1 ?
longest-idle longest idle hunting
peer peer hunting
sequential sequential hunting
r5uk(config-ephone-hunt)#?
EPHONE-HUNT configuration commands:
auto enable automatic features
default Set a command to its defaults
exit Exit from ephone hunt configuration mode
final final number for hunt group
list list of number in hunt group
no Negate a command or set its defaults
no-reg not register pilot number to gatekeeper
pilot pilot number for hunt group
preference preference of pilot number
statistics enable statistic information collect
timeout timeout in seconds for hunting
r5uk(config-ephone-hunt)#
VoiceBootcamp
60
• Pilot - Defines the pilot number, which is the number that callers dial to reach the hunt group.
• List - Defines the list of numbers to which the ephone hunt group redirects the incoming calls.
There must be between two and twenty numbers in the list.
• Final - Defines the last number in the ephone hunt group, after which the call is no longer
redirected. This number can be an ephone-dn primary or secondary number, a voice-mail pilot
number, a pilot number of another hunt group, or an FXS number.
• Each hunt group can consist of 20 ephone-dn as members
• Each hunt group can have a final destination where if no members answer the call, call can be
redirected to final destination.
Note
Once a final number is defined as a pilot number of another hunt group, the pilot number of the first
hunt group cannot be configured as a final number in any other hunt group.
For more information please visit www.cisco.com

CCME Hunting
Inbound call to 6500
Ephone-hunt 1 seq
pilot 6500
09:00 06/500/05
list 6001, 6002 6001
final 6000 6001
timeout 5 IP phone 1
ephone-hunt 2 peer VoiceBootcamp Inc.
pilot 6000 If 6001 is busy
list 6002, 6001, 6003 and/or not
final 3001  can not be 6500 answering
preference 1
timeout 30
no-reg
09:00 06/500/05
6001
6001
VoiceBootcamp
IP phone 2
VoiceBootcamp Inc.
61
• First hunt-group
•If user dial 6500 call will first go to 6001. If 6001 is busy and/or not answering then call
will be forwarded to 6002
• Second hunt-group
•If the last call that answer was 6001 then if some one dial 6000 call will go to 6003.
CallManager Express
DN overlays
• Assign up to 25 ephone-dn to a single phone button

• Call Waiting is not allowed in overlay functions.
• Use Advanced Algorithm
• Overlaid ephone-dns can use ephone-dns with the same
number or different numbers.
• If a phone is using an overlaid ephone-dn on an active
call, call waiting will be disabled for any incoming calls to
any ephone-dn in the overlay set.
VoiceBootcamp
62
• Overlaid ephone-dns allow more than one ephone-dn to share the same physical line button on
an IP phone.
• Overlaid ephone-dns can be used to receive incoming calls and place outgoing calls. Up to 25
ephone-dns can be assigned to a single phone button.
• If a phone is using an overlaid ephone-dn on an active call, call waiting will be disabled for any
incoming calls to any ephone-dn in the overlay set.
CCME DN overlay Example
ephone-dn 10
06/500/05 6001 number 6601
6001 no huntstop
preference 0
6601
UK phone 1 ephone-dn 11
number 6601
Cisco CME no huntstop
preference 1
06/500/05 6002
Ephone-dn 12
6002 number 6601
6601 huntstop
UK phone 2 preference 5
ephone 1
Cisco CME
mac-address 111.111.111
06/500/05 6001 button 1:1 2o10,11,12
6002 ephone 1
6601 mac-address 111.111.112
UK phone 3 button 1:2 2o10,11,12
VoiceBootcamp
ephone 1
Cisco CME
mac-address 111.111.113 63
button 1:2 2o12 11 10
The following example creates 3 lines (ephone-dns) that are shared across a IP phones to handle 3
simultaneous calls to the same telephone number. 3 instances of a shared line with the extension
number 6601 are overlaid onto a single button on phones. A typical call flow is as follows. The first
call goes to ephone 1 (highest preference) and rings button 1 on all phones (huntstop is off).
The call is answered on ephone 1. A second call to extension 6601 hunts onto ephone-dn 2 and
rings on the two remaining ephones, 2 and 3. The second call is answered by ephone 2. A third
simultaneous call to extension 6601 hunts onto ephone-dn 3 and rings on ephone 3, where it is
answered. Note that the no huntstop command is used to allow hunting for the first two ephone-
dns, and the huntstop command is used on the final ephone-dn to stop call- hunting behavior. The
preference command is used to create different selection preferences for each
ephone-dn.
CallManager Express Shared DN
overlay Example
09:00 06/500/05 09:00 06/500/05

6001 6002
6001 6002
IP phone 1 3001 IP phone 2 3001
Cisco CME Cisco CME
ephone-dn 10
ephone-dn 1 number 3001 ephone-dn 2
number 6001 number 6002
ephone-dn 11
ephone 1 number 3002 ephone 2
mac-address 5001.5001.5001 mac-address 2222.2222.2222
button 1:10,11,12 ephone-dn 12 button 1:2 2o1,11,12
number 3003
Overlay sets can be shared across multiple phones
VoiceBootcamp
64
Restrictions
• Ephone-dn overlays disable call waiting.
• If a phone is using an overlaid ephone-dn on an active call, call waiting will be disabled for any
incoming calls to any ephone-dn in the overlay set.
Callmanager Express
System Message
• Allows you to change the default message on the IP

Phone
telephony-service
system message “Welcome to iNet?!”
09:00 06/5/07 6001
6001
Welcome to iNet?!
65
VoiceBootcamp
• Define a system messages such as company name or department name etc.
CME Extension Mobility
•Allow user to login to a physical other than their own phone
•Sales per going to remote branch office can login to one of the phone in BR office.
Extension movies with the user
•Usually known as Follow Me Number
•User must login and logout to use EM Features
•Some company use EM permanent solution to authenticate users
Perform the following tasks to enable Extension Mobility in Cisco Unified CME:
• Configuring Cisco Unified CME for Extension Mobility
• Configuring a Logout Profile for an IP Phone
• Enabling an IP Phone for Extension Mobility
• Configuring a User Profile
VoiceBootcamp
66
A user login service allows phone users to temporarily access a physical phone other than their own
phone and utilize their personal settings, such as directory number, speed-dial lists, and services, as if
the phone is their own desk phone. The phone user can make and receive calls on that phone using the
same personal directory number as is on their own desk phone
To create a logout profile to define the default appearance for a Cisco Unified IP phone that is enabled
for Extension Mobility
Configuring Cisco Unified CME for
Extension Mobility
Router (config) telephony-service

Router(config-telephony)# url authentication
http://192.168.1.198/CCMCIP/authenticate.asp secretname psswrd
authentication credential application-name password
em keep-history
em logout 19:00 24:00
VoiceBootcamp 67
Router(config-telephony)# url authentication http://192.0.2.0/CCMCIP/authenticate.asp secretname

psswrd
Instructs phones to send HTTP requests to the authentication server and specifies which credential to
use in the requests.
This command is supported in Cisco Unified CME 4.3 and later versions. Required to support
Automatic Clear Call
history. URL for internal authentication server in Cisco Unified CME is http://CME IP
Address/CCMCIP/authenticate.asp.
authentication credential application-name password

Creates an entry for an application's credential in the database used by the Cisco Unified CME
authentication server.
EM keep-history
Specifies that Extension Mobility will keep, and not automatically clear, call histories when users log out
from Extension Mobility
phones
em logout 8:00 24:00

Defines up to three time-of-day timers for automatically logging out all Extension Mobility users.
VoiceBootcamp
Configuring a Logout Profile for an IP
Phone
To create a logout profile to define the default appearance for a Cisco Unified IP phone that
is enabled for Extension Mobility
voice logout-profile 1
user name password password
number 3002 type beep-ring
speed-dial 2 5002 blf
Pin 1234
VoiceBootcamp 68
Configuring a Logout Profile for an IP
Phone
To create a logout profile to define the default appearance for a Cisco Unified IP phone that
is enabled for Extension Mobility
Pin 1234
VoiceBootcamp 69
Enabling an IP Phone for Extension
Mobility
To enable the Extension Mobility feature on an individual Cisco Unified IP phone in Cisco Unified
CME,
Pin 1234
Ephone 1
mac-address Y.Y.Y.Y
button 1:1
type 7961
logout-profile 11
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70
All SCCP Cisco Unified IP phones with displays that support URL provisioning for Feature buttons are
supported by Extension Mobility, including the Cisco Unified Wireless IP Phone 7920, Cisco Unified
Wireless IP Phone 7921, and Cisco IP Communicator.
Configuring a User Profile
To enable the Extension Mobility feature on an individual Cisco Unified IP phone in Cisco Unified
CME,
voice user-profile 1
pin 12345
user me password pass123
number 5001 type silent-ring
number 5003 type feature-ring
number 5004 type monitor-ring
number 5005,5006 type overlay
number 5007,5008 type cw-overly
speed-dial 1 3001 speed-dial 2 3002 blf
VoiceBootcamp
71
All SCCP Cisco Unified IP phones with displays that support URL provisioning for Feature buttons are
supported by Extension Mobility, including the Cisco Unified Wireless IP Phone 7920, Cisco Unified
Wireless IP Phone 7921, and Cisco IP Communicator.
Configuring Transcoding in IOS
voice-card 1 dspfarm profile 1 transcode
dsp services dspfarm codec g711ulaw
codec g711alaw
sccp local FastEthernet 0/1.101 codec g729ar8
sccp codec g719abr8
sccp ccm 135.Y.67.240 identifier 1 maximum sessions 6
associate application sccp
sccp ccm group 123
associate ccm 1 priority
telephony-service
associate profile 1 register R1MTP
ip source-address 10.5.49.500 port 5000
keepalive retries 5
sdspfarm units 1
switchover method immediate
sdspfarm transcode sessions 40
switchback method immediate
sdspfarm tag 1 R1MTP
switchback interval 5
VoiceBootcamp 72
Transcoding compresses and decompresses voice streams to match endpoint-device capabilities.

Transcoding is required when an incoming voice stream is digitized and compressed (by means of a
codec) to save bandwidth, and the local device does not support that type of compression
When do you need Transcoding?

• Ad hoc conferencing—One or more remote conferencing parties uses G.729.
• Call transfer and forward—One leg of a Voice over IP (VoIP)-to-VoIP hairpin call uses G.711 and the
other leg uses G.729. A hairpin call is an incoming call that is transferred or forwarded over the same
interface from which it arrived.
• Cisco Unity Express—An H.323 or SIP call using G.729 is forwarded to Cisco Unity Express.
• Cisco Unity Express supports only G.711, so G.729 must be transcoded.
• Music on hold (MOH)—The phone receiving MOH is part of a system that uses G.729. The G.711 MOH
is transcoded into G.729 resulting in a poorer quality sound due to the lower compression of G.729
Presence with CME
• Watch the status of another user in your directory

• Presence enables the calling party to know before dialing whether the called
party is available
• Presence uses SIP SUBSCRIBE and NOTIFY methods to allow users and
applications to subscribe to changes in the line status of phones in a Cisco
Unified CME system
• Presence supports Busy Lamp Field (BLF) notification features for speed-
dial buttons and directory call lists for missed calls, placed calls, and received
calls.
VoiceBootcamp
73
Presence Configurations
Enable Presence in CME Enters SIP user-agent configuration mode
to configure the user agent.
Configure terminal
sip-ua
Allows the router to accept incoming
presence
presence requests
Presence Enables presence service and enters presence
Max-subscriber 128 configuration
Presence call-list mode.
Enables presence service and enters presence

configuration mode.
Enabling a Directory Number Globally enables BLF monitoring for
to be Watched directory numbers in call lists and
directories on all locally registered phones
configure terminal
VoiceBootcamp
ephone-dn 1 or voice register dn 1
number 6001
allow-watch  allow extenion to be watched
74
To enable a line associated with a directory number to be monitored by a phone registered to a Cisco
Unified CME router, perform the following steps. The line is enabled as a presentity and phones can
subscribe to its line status through the BLF call-list and BLF speed-dial features. There is no restriction on
the type of phone that can have its lines monitored; any line on any IP phone or on an analog phone on
supported voice gateways can be a presentity.
configure terminal
ephone-dn 1 or voice register dn 1
number 6001
allow-watch  allow extenion to be watched
NOTE: voice register is used for SIP IP phone.

Presence on CME Speed Dial
•Watcher can see the status of a internal as well as external number

•Using BLF Speed Dial to monitor the status of another extension
Ephone 1
mac-address x.x.x.x
button 1:1
blf-speed-dial 1 6002 label Peter Smith
presence call-list
Voice register pool 1

id mac-address x.x.x.x
number 1 dn 1
blf-speed-dial 1 6002 label Peter Smith
presence call-list
VoiceBootcamp 75
Blf-speed-dial is a special speed dial that can track the status of the destination device.
NOTE: presence call-list is used to ensure that if this speed number 6002 shows up in a directory list then
presence status should be visible
Single Number Reach in CME
• Answer incoming calls on their desktop IP phone or at a

remote destination, such as a mobile phone
• Pick up in-progress calls on the desktop phone or the
remote phone without losing the connection
• Send calls to remote device and pull call back from remote
device using Resume softkey
VoiceBootcamp 76
The Single Number Reach (SNR) feature allows users to answer incoming calls on their desktop IP
phone or at a remote destination, such as a mobile phone, and to pick up in-progress calls on the desktop
phone or the remote phone without losing the connection. This allows callers to use a single number to
reach the phone user. Calls that are not answered can be forwarded to voice mail
Single Number Reach restriction in CME

• Each IP phone supports only one SNR directory number
SNR feature is not supported for the following:
–SIP phones or SCCP-controlled analog FXS phones.
–MLPP calls.
–Secure calls.
–Video calls.
–Hunt group directory numbers (voice or ephone).
–MWI directory numbers.
–Trunk directory numbers.
• An overlay set can support only one SNR directory number and that directory number must be the
primary directory number.
• Call forward no answer (CFNA), configured with the call-forward noan command, is disabled if SNR
is configured on the directory number. To forward unanswered calls to voice mail, use the cfwd-
noan keyword in the snr command
• If the SNR directory number is the transferred number (Xee) in a blind or consultive transfer, the user
cannot send the call to the remote phone.
• When an SNR call is answered on the remote phone and the call is then transferred, parked, or joined
in a hardware conference in Cisco Unified CME, the user cannot resume the call on the desktop IP
phone.
VoiceBootcamp
Single Number Reach in CME
ephone-template 1
softkeys idle Dnd Gpickup Pickup Mobilit
softkeys connected Endcall Hold LiveRcd Mobility
ephone-dn 10
number 6001
mobility
Snr 4163013001 3 delay 5 timeout 15 cfwd-noan 6600
VoiceBootcamp 77
The Single Number Reach (SNR) feature allows users to answer incoming calls on their desktop IP
phone or at a remote destination, such as a mobile phone, and to pick up in-progress calls on the desktop
phone or the remote phone without losing the connection. This allows callers to use a single number to
reach the phone user. Calls that are not answered can be forwarded to voice mail
Single Number Reach restriction in CME

• Each IP phone supports only one SNR directory number
SNR feature is not supported for the following:
–SIP phones or SCCP-controlled analog FXS phones.
–MLPP calls.
–Secure calls.
–Video calls.
–Hunt group directory numbers (voice or ephone).
–MWI directory numbers.
–Trunk directory numbers.
• An overlay set can support only one SNR directory number and that directory number must be the
primary directory number.
• Call forward no answer (CFNA), configured with the call-forward noan command, is disabled if SNR
is configured on the directory number. To forward unanswered calls to voice mail, use the cfwd-
noan keyword in the snr command
• If the SNR directory number is the transferred number (Xee) in a blind or consultive transfer, the user
cannot send the call to the remote phone.
• When an SNR call is answered on the remote phone and the call is then transferred, parked, or joined
in a hardware conference in Cisco Unified CME, the user cannot resume the call on the desktop IP
phone.
VoiceBootcamp
Voice Gateways and Protocols
VoiceBootcamp 78
Voice Gateway Protocols
H323 Gateway Other gateways/Trunk
• SIP Trunk
• Gatekeeper Trunk
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79
Gateways provide a methods for connecting an IP telephony network to the Public Switched Telephone
Network (PSTN), a legacy PBX, or key systems.
Cisco access gateways allow Cisco Unified CallManager to communicate with non-IP
telecommunications devices
Cisco Unified CallManager supports the following gateway protocols:

•H.323
•Peer to Peer protocol
•No central control
•Each gateway act on its own
•Dial plan and translation can be configured per gateway basis.
•Media Gateway Control Protocol (MGCP)

• Centralized Dial Plan and Administration
• Call Agent in charge of the gateway
• master/slave relationship
•Gatekeeper
•Design to provide a centralize gateway, bandwidth and dial plan management
for h323 gateways.
•Gateway must register to the gatekeeper before they can route calls
Digital Voice Signaling: ISDN-PRI
ISDN Q931
ISDN Q921 PSTN
E1 Framing
isdn switch-type primary-ni

! Globally defines isdn switch type
controller E1 0/0
framing no-crc4
linecode hdb3 Defines T1-PRI under the T1 controller
pri-group timeslots 1-24
!
int s0/0:15 D-channel (int s0/0:23) and voice-port
isdn incoming-voice voice will be automatically created once pri-
isdn switch-type primary-ni group is defined on the T1 controller.
! D-channel carries the call information
voice-port 0/0:15 such as DNIS (called number) and
VoiceBootcamp
cptone GB ANI (calling number)
!
dial-peer voice 1 pots
destination-pattern 9.T
incoming called-number . Create pots dial-peer which defines
direct-inward-dial voice call routing rules
port 0/0:15
80
• ANI: Automatic Number Identification, a.k.a Calling number

• DNIS: Dialed Number Identification Service, a.k.a called number
VoIP Signaling Protocols
CallManager
PSTN
VoIP
Signaling:
H.323
MGCP
Gatekeeper
VoiceBootcamp
SIP Gateway
81

•H.323
•Media Gateway Control Protocol (MGCP)

• Centralized Dial Plan and Administration
• Call Agent in charge of the gateway
• master/slave relationship
•Gatekeeper
•Design to provide a centralize gateway, bandwidth and dial plan management
for h323 gateways.
•Gateway must register to the gatekeeper before they can route calls
H.323 Gateway
PSTN IP
PRI Layer 3
PSTN
Layer 2 H.225 and H.245 over TCP

Framing
Cisco CallManager
• H.323 is a “peer-to-peer” protocol

• All PSTN signaling terminates on gateway
• H.225 and H.245 signaling communications over TCP between
gateways and CallManager
• Media over UDP directly between gateways and IP phones: CCM
responsible for call setup/tear-down and capability negotiation only
VoiceBootcamp
• Gateway status in CCM always remain “Unknown”
82

•H.323
Advantage of H323 Gateway

•Protocol of choice for distributed architecture
•More control over gateway and call routing
Disadvantage of h323 gateway

•No centralize management
Basic H.323 IOS Configuration
Defines T1-PRI as PSTN signaling
controller T1 1/0
framing esf D-channel and its configurations
linecode b8zs
pri-group timeslots 1-24 Dial Peer for VoIP Leg
!
interface Serial1/0:23
isdn switch-type primary-ni
isdn incoming-voice voice Destination-pattern for digit
! matching
dial-peer voice 1 voip Session target pointing to ip
destination-pattern 3... address of remote H.323 peer: i.e.
session target ipv4:135.XX.100.12 Call Manager’s IP addr.
codec g711ulaw
dtmf-relay h245-alphanumeric Use g711u codec. Default is g729
!
dial-peer voice 9 pots Enables DTMF relay using H245-
destination-pattern 9T alpha. Default is disabled
direct-inward-dial
incoming called-number .T Pots dial-peer pointing to the PRI with
VoiceBootcamp
port 1/0:23 destination-pattern, pots peers strips
explicitly matched digit(s) in
destination-pattern
83
Controller T1
• T1 parameters must be provided by the telco.
• ISDN Switch type must be set properly
• If linecode and/or framing is not configured properly, Controller will generate Layer 2 Alarm.
Dial Peer
• Two type of dial peer
•POT
•POT dial peer points call to PSTN and/or analog network
•VOIP
•VOIP dial peer points the call to another voip network such as gateway or
CallManager
Destination-pattern 9T
•Pattern used to match outbound call
Direct-inward-dial
•Allow the call to pass through the router and find a best possible destination pattern
•Usually used to match DID and/or route calls to specific number
Incoming called-number .T
• match any inbound calls to a particular dial peer
Additional H.323 IOS
Configuration Options
interface loopback 0 Forces this gateway to use the loopback
ip address XX.33.33.33 255.255.255.0 interface for all H.323 signal and UK
h323-gateway voip interface
h323-gateway voip bind srcaddr XX.33.33.33 traffic.
!
voice class h323 1 H.225 setup redundancy: try a second
h225 timeout setup 5 voip dial-peer if the remote H.323 peer
! does not response in 5 seconds.
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw H.245 codec negotiation flexibility:
! negotiate to g729 if possible; otherwise
dial-peer voice 1 voip g711ulaw is okay too.
destination-pattern 3...
session target ipv4:135.XX.100.12 Try this dial-peer first if 3… is match
voice-class h323 1 because it has the highest preference:
voice-class codec 1 0. Default preference value, therefore
! invisible in dial-peer configuration.
dial-peer voice 2 voip
destination-pattern 3...
VoiceBootcamp
session target ipv4:135.XX.100.11 If the IP host in dial-peer 1
voice-class h323 1 (135.XX.100.12) does not response
voice-class codec 1 H.225 setup in 5 seconds, try this dial-
preference 1 peer as it has lower preference.
84
In order for Cisco router to function as a h323 gateway, it is suggested that you configure the H323
bind interface.
H323 bind interface basically advice the router to source all traffic from a particular IP address in
this case the loopback 0
When a voip call is made to a destination IP address, often network congestion can delay the call
establishment. In order to fine tune a voice network, it may be necessary to provide a fault tolerant
solution by providing a backup connection.
Voice Class H323 allows you to reduce the h225 time so that call leg does not wait for too long for a
remote gateway to response. If originating gateway does not get response within configured
interval then move to the next dial peer
Voice Class Codec allows you to select multiple codec and it is attached to dial peers.
Default codec is: G.729

Call Manager H.323 Gateway
Configuration
2a
VoiceBootcamp 85
NOTE:
Device Name: is either IP address of the bind src address from the router or FQDN that mapped
to the IP address of bind src address
Registration Status will always be unknown. Only way to verify if it is registered in CallManager
or not, if look for IP Address: If it shows the correct IP address then configuration is fine.
Define the appropriate device pool. If this gateway belong to a site that has location defined
(location will be covered later) then you must select location here as well.
Media Termination Point Require must be check if remote gateway is a h323v1

Call Manager H.323 Gateway Config. (cont’d)
2b Continued from CCM H.323 Gateway Configuration Page:
VoiceBootcamp 86
Signifcant Digit – Advice callmanager how many digit to strip off from the incoming call number before
looking for a match.
Incoming call to CallManager with number 14163133001 with significant digit set to 4 means CallManager
will take the last 4 digit in this case 3001 and discard the remain digit before finding a phone to ring.
Redirecting Number IE delivery - accept the Redirecting Number IE in the incoming SETUP message to
the Cisco CallManager.
H323 Dial Peer
7 Digit (with 9 access code) 11 (long distance)

dial-peer voice 7 pots dial-peer voice 11 pots
destination-pattern 9[2-9]…… destination-pattern 91[2-9]..[2-9]……
forward-digits 7 forward-digits 11
port 1/0:23 port 1/0:23
911 calls Overseas or international

Dial-peer voice 911 pots Dial-peer voice 111 pots
Destination-pattern 911 Destination-pattern 9011T
Forward-digits 3 Port 1/0:23

Prefix 011
Port 1/0:23
VoiceBootcamp
87
• Any explicit match will be discarded

destination-pattern 91[2-9]..[2-9]……
forward-digits 11
port 1/0:23
If user dial 914168392727 the resulting number will be 4168392727 before it reach PSTN.
However since we are saying forward-digit 11 that means we are instructing the router to send the
last 11 digit of the dialed number. So the number that reach the PSTN IS 14168392727
When you are not sure how many digit to forward, then use prefix to send what ever the digit you
need to send in order to complete the call.
MGCP (Media Gateway Control
Protocol)
• Media Gateway (MG) contains “simple” endpoints,

which can be either analog voice-ports (FXS/FXO) or
digital (T1-PRI/T1-CAS) voice trunks
• Call Intelligence of these endpoints are provided by
Media Gateway Controller (MGC) or Call Agent (CA),
in our case, the Call Manager
• Master/Slave relationship between MGC/CA and MG
• MGCP messages are sent over IP/UDP between MGC
and MG - Signaling Plane
• Voice traffic is carried over IP/UDP
VoiceBootcamp 88
• The endpoints can be physical or virtual. Devices like an IP phone and gateway are endpoints.
• In VG100, each Foreign Exchange Station/ Foreign Exchange Office (FXS/FXO) port are
endpoints.
• MGCP consists of eight commands:

• RQNT – NotificationRequest: CallManager can issue a NotificationRequest command to a
gateway, instructing the gateway to watch for specific events such as hook actions or Dual-Tone
Multifrequency (DTMF) tones on a specified endpoint. RQNT is also used to request a gateway
to apply a specific signal to endpoint (i.e. dial tone, ringback, etc).
• NTFY – Notify: The gateway uses the Notify command to inform the CallManager when the
requested events occur.
• CRCX – CreateConnection: CallManager uses the CreateConnection command to create a
connection that terminates in an endpoint inside the gateway.
• MDCX – ModifyConnection: CallManager uses the ModifyConnection command to change
the parameters associated to a previously established connection.
• DLCX – DeleteConnection: CallManager uses the DeleteConnection command to delete an
existing connection. The DeleteConnection command may also be used by a gateway to
indicate that a connection can no longer be sustained.
• AUEP – AuditEndpoint: CallManager uses the AuditEndpoint commands to audit the status of
an endpoint associated with it.
• AUCX – AuditConnection: CallManager uses the AuditConnection commands to audit the
status of any connection associated with it.
• RSIP – RestartInProgress: The gateway uses the RestartInProgress command to notify the
CallManager that the gateway, or a group of endpoints managed by the gateway, is being taken
out of service or is being placed back in service. There are three types of restart:
• Restart – endpoint in service; Graceful – wait until call clearing; Forced – endpoint out of service.
VoiceBootcamp
IOS MGCP PRI Backhaul Configuration
hostname rXsfo
!
Must match “Domain Name” on
mgcp MGCP Gateway page on CCM
mgcp call-agent 135.XX.100.11
mgcp bind control source looopbac0
mgcp bind media source loopback0 Enables MGCP process globally
!
ccm-manager redundant-host 135.XX.100.12 Defines Primary Call-agent: the ip
ccm-manager mgcp address of primary CCM
ccm-manager fallback!
Defines secondary call-agent
controller T1 1/0
linecode b8zs
MGCP version 0.1 with
framing esf
CallManager
pri-group timeslots 1-7 service mgcp
!
interface Serial1/0:23 Defines on the T1 controller that
no ip address the PRI ports will be serviced by
no logging event link-status MGCP
isdn incoming-voice voice
isdn bind-l3 ccm-manager Under D-channel, binds L3 (Q.931)
! to call manager
VoiceBootcamp
service mgcpapp Defines MGCP as the call
port 1/0:23 application under pots dial-peer
89
NOTE
It is often a good idea to bind MGCP traffic to a reliable interface such as Loopback or VLAN 10X
interface.
Do not forget to include service mgcp command in controller
Under serial interface, isdn bind-l3 command is a important. Ensure it is there, it basically bind the
D channel to the CallManager
MGCP: Call Manager Configuration
2
Must match with hostname
and ip domain-name (if
applicable) on the IOS
gateway
VoiceBootcamp
90
When adding MGCP gateway, you must know the name of your router. Also if ip domain-name is
configured with domain name such as cisco.com then MGCP Domain name will be hostname.cisco.com
Once domain name is defined, define the slot where Voice module is in. Based on that the Call Manager
will know which Voice port to control
MGCP: Call Manager Configuration
(cont’d)
VoiceBootcamp
91
In Gateway Configuration Ensure that Channel Selection Order is set correctly. Often if you do a debug
and noticed that you are getting an error message of channel and/or circuit not available it is possible that
channel selection order is causing such issue.
Useful IOS MGCP Verification Commands
GW1#sh isdn stat

Global ISDN Switchtype = primary-ni
ISDN Serial1/0:23 interface
dsl 0, interface ISDN Switchtype = primary-ni
L2 Protocol = Q.921 L3 Protocol(s) = CCM-MANAGER
Layer 1 Status:
ACTIVE
Layer 2 Status:
TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED
Layer 3 Status:
0 Active Layer 3 Call(s)
Active dsl 0 CCBs = 0
The Free Channel Mask: 0x8000003F
Number of L2 Discards = 2, L2 Session ID = 30
Total Allocated ISDN CCBs = 0
VoiceBootcamp 92
When you type show isdn status in MGCP router, Layer 2 Status will be multiple frame established
only when CCM is registered.
SIP Gateway
• SIP is a session initiated protocol

• SIP uses a request/response method to establish
communications
• Identification of users in a SIP network works through
• A unique phone or extension number.
• A unique SIP address that appears similar to an e-mail
address and uses the format sip:<userID>@<domain>.
• A signaling interface (trunk) must be configured to
receive/send calls.
VoiceBootcamp 93
• A SIP network uses the following components:

• SIP Proxy Server—The proxy server works as an intermediate device that receives
SIP requests from a client and then forwards the requests on the client's behalf. Proxy
servers can provide functions such as authentication, authorization, network access
control, routing, reliable request retransmission, and security.
• Redirect Server—The redirect server provides the client with information about the
next hop or hops that a message should take, and the client then contacts the next hop
server or user agent server directly.
• Registrar Server—The registrar server processes requests from user agent clients for
registration of their current location. Redirect or proxy servers often contain registrar
servers.
• User Agent (UA)—A combination of user agent client (UAC) and user agent server
(UAS) that initiates and receives calls. A UAC initiates a SIP request. A UAS is a server
application that contacts the user when it receives a SIP request. The UAS then returns
a response on behalf of the user. Cisco CallManager can act as both a server or client
(a back-to-back user agent).
• SIP uses a request/response method to establish communications between various components
in the network and to ultimately establish a call or session between two or more endpoints. A
single session may involve several clients and servers.
• Identification of users in a SIP network works through
• A unique phone or extension number.
• A unique SIP address that appears similar to an e-mail address and uses the format
sip:<userID>@<domain>. The user ID can be either a user name or an E.164 address.
Cisco CallManager only supports E.164 addresses; it does not support email addresses.
VoiceBootcamp
SIP Gateway (cont’d)
• SIP signaling interfaces connect Cisco CallManager

networks and SIP networks
• SIP signaling interfaces use port-based routing
• Cisco CallManager accepts calls from any SIP device as long
as the SIP messages arrive on the configured incoming port
• Cisco CallManager requires an RFC 2833 dual tone
multifrequency (DTMF) compliant MTP device to make SIP
calls
• MTP is required since SIP use in-band and SCCP phone use
out-band
VoiceBootcamp 94
SIP and CallManager Connectivity
All protocols require that either a signaling interface (trunk) or a gateway be created to accept and
originate calls. For SIP, the user must create a signaling interface
SIP signaling interfaces connect Cisco CallManager networks and SIP networks that are served by a
SIP proxy server.
SIP signaling interfaces use port-based routing, with one SIP signaling interface connecting to a SIP
network. Cisco CallManager accepts calls from any SIP device as long as the SIP messages arrive on
the configured incoming port. When configuring multiple signaling interfaces, configure a unique
incoming port for each SIP interface. Use of the same port as an incoming port for multiple signaling
interfaces causes an alarm
Media Termination Point (MTP) Devices

Cisco CallManager requires an RFC 2833 dual tone multifrequency (DTMF) compliant MTP device to
make SIP calls. The current standard for SIP uses in-band Real-Time Transport Protocol (RTP) payload
types to indicate DTMF tones. AVVID components such as SCCP IP phones, support only out-of-band
DTMF payload types. Thus, an RFC 2833 compliant MTP device acts as a translator between inband
and out-of-band DTMF.
Basic Outgoing Call
You can initiate outgoing calls to a SIP device from any Cisco CallManager device. A
Cisco CallManager device includes SCCP IP Phones or fax devices that are connected to
Foreign Exchange Station (FXS) gateways. For example, an SCCP IP Phone like 7960 with SIP
Image can place a call to a SIP endpoint. The SIP device answering the call triggers media
establishment.
Basic Incoming Call

Any device on the SIP network, including SIP IP Phones or fax devices that are connected to
FXS gateways can initiate incoming calls. For example, a SIP endpoint like SIPURA can initiate a
call to an SCCP IP Phone. The SCCP IP Phone answering the call triggers media establishment.
Use of Early Media

While the PSTN provides inband progress information to signal early media (such as a ring tone
or a busy signal), the same does not hold true for SIP. The originating party includes Session
Description Protocol (SDP) information, such as codec usage, IP address, and port number, in
the outgoing INVITE message. In response, the terminating party sends its codec, IP address,
and port number in a 183 Session Progress message to indicate possible early media.
The 183 Session Progress response indicates that the message body contains information about
the media session. Both 180 Alerting and 183 Session Progress messages may contain SDP,
which allows an early media session to be established prior to the call being answered.
When early media needs to be delivered to SIP endpoints prior to connection,
Cisco CallManager always sends a 183 Session Progress message with SDP. While
Cisco CallManager does not generate a 180 Alerting message with SDP, it does support the 180
Alerting message with SDP when it receives one
VoiceBootcamp
SIP-Initiated Call Transfer
Cisco CallManager does not support SIP-initiated call transfer and does not accept receiving
REFER requests or INVITE messages that include a Replaces header. When Cisco CallManager
receives a REFER request, it returns a 501 Not Implemented message. When
Cisco CallManager receives an INVITE message with a Replaces header, it processes the call
and ignores the Replaces header.
SIP Gateway Call
PSTN IP
Cisco CallManager
IP
MTP
94168391717
VoiceBootcamp 96
Forwarding a DTMF Calls
1. The SIP Phone initiates a payload type response when the user enters a number on the keypad.
The SIP Phone transfers the DTMF in-band digit (per RFC 2833) to the MTP device.
2. The MTP device extracts the in-band DTMF digit and passes the digit out of band to
Cisco CallManager.
3. Cisco CallManager then relays the DTMF digit out of band to the gateway or IVR system
SIP Gateway Configurations
voice service voip Allow PSTN calls (h323) to reach
allow-connections h323 to sip SIP network
Dial-peer voice 1 pots

Application session defines that standard session
application sessions
application will be invoked for this dial peer
destination-pattern 9T
incoming called-number .
direct-inward-dial
port 1/0:23
dial-peer voice 101 voip Define a VoIP Dial-peer to send calls to CCM
destination-pattern [35]… Must change the protocol to SIPV2 as default is h323
session protocol sipv2
session target ipv4:135.11.100.11 For Cisco IP phone to work, you must use SIP-
dtmf-relay sip-notify NOTIFY As DTMF Relay
codec g711ulaw
VoiceBootcamp 97
H.323-to-H.323: By default, H.323-to-H.323 connections are disabled and POTS-to-any and any-to
POTS connections are enabled.
SIP Trunk in CallManager
VoiceBootcamp
98
From Device Menu, click on Trunk. Then add a new trunk based on SIP Trunk.
Device Name: enter meaning full device name.
Media Termination Point must be check since SCCP IP phone does not understand SIP inband
DTMF.
Therefore you must have an MRGL applied to this trunk with MTP in it
Destination Address – IP address of the SIP Gateway
Incoming Port – 5060 – SIP Gateway must send calls to this port. CCM use port based routing.
Voice Translation
• Voice Translation Profiles introduce a new scheme to

translate numbers
• Translation rule can have up to 16 sub – rule matched in
orderly fashion. First rule match, subsequent rules are
ignored.
• Translation profile is used to apply the rule to calling,
called or redirected number
• Translation rule can be applied either at Outgoing or
Incoming direction.
• It can be applied to voice port, dial peer, trunk group,
source IP group, NFAS interface
VoiceBootcamp 99
• Voice Translation Profiles introduce a new scheme to translate numbers. The older translation
rules are to be gradually phased out of the system. Cisco strongly recommends you only use one
scheme of translation rules
Translation
Example 1
This example replaces the first occurrence of the number "123“ with "456".
voice translation-rule 1
rule 1 /123/ /456/
These are test voice translation-rule examples:

rXuk#test voice translation-rule 1 123
Matched with rule 1
Original number: 123 Translated number: 456
VoiceBootcamp 100
Translation cont’d
Example 2
This example shows how to replace any occurrence of "123" at the start
of a number with "456".
rule 1 /^123/ /456/
These are test voice translation-rule examples.
Matched with rule 1
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Matched with rule 1
101
Voice Translation Profile.
• voice translation-rule 1
rule 1 /1#4402/ /9/
rule 2 /1#440/ /90/
• voice translation-profile ChangeDNIS

translate called 1
• Voice-port 1/0:23
translation-profile outgoing ChangeDNIS
VoiceBootcamp 102
• In order to create voice translation rule first create rule that you want to match against incoming
or outgoing call.
• Once rules are created you must attached that rule to a profile such as ChangeDNIS. When
applying the rule to a profile, you going to define what number you want to modify, called or
calling or redirected number.
• Once rule has been defined in profile, apply the profile to the voice port or where ever you want
to apply this translation rule. Either apply this incoming or outgoing direction.
Translation Cont’d
Wildcard Definition
. Any single digit
0 to 9,*,# Any specific character
[0-9] Any range or sequence of characters
* Modifier—match none or more occurrences
+ Modifier—match one or more occurrences
VoiceBootcamp 103
• Various wildcard can be used to construct your match pattern.

Wildcard Combination Definition
Any digit followed by none or more ocurrences. This is

.*
effectively anything, including null.
Any digit followed by one or more ocurrences. This is effectively

.+
anything, except null.
^$ No digits, null
VoiceBootcamp 104
Some example of wildcard usages

Example 2
This example replaces all numbers with "5554000".
rule 1 /.*/ /5554000/
Matched with rule 1
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• Replace all numbers with 5554000
105
• Useful for changing caller ID for a company to a specific number

Example 3
This example replaces all numbers, except null, with "5554000".
voice translation-rule 2 rule 1 /.+/ /5554000/
router#test voice translation-rule 2 123
Matched with rule 1
router#test voice translation-rule 2 ""

Didn't match with any of rules
VoiceBootcamp 106
• Replace all number except empty string.

ISDN D Channel
• R1sfo(config) interface serial 1/0:23

• r1sfo(config-if)#isdn ?
– bchan-number-order Specify bchannel starting number order
– bind-l3 Bind Layer 3 protocol to signaling interface
– caller Specify incoming telephone number to be verified
– calling-number Specify Calling Number included for outgoing calls
– outgoing Options for outgoing IEs and messages
– outgoing-voice Specify information transfer capability for voice calls
– overlap-receiving Specify if the interface will do Overlap Receiving
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• bchan-number-order – used this to change the channel selection order either top down or bottom
107
up
• Bind-l3 – bind layer 3 address to CCM
• Calling-number – Override the caller ID of outgoing calls.
• Outgoing – define various options for outgoing IEs
Gatekeeper
• Gatekeeper is design to provide a centralized call

management for H323 network
• Gateway registers to gatekeeper before routing calls
• Each gateway may advertise all the prefix it can serve
• Gatekeeper use Tech Prefix, Zone Prefix and Alias to route
calls
• Gateway will use gatekeeper for centralize dial plan and CAC
• Gatekeeper can provide bandwidth control
VoiceBootcamp 108
Cisco gatekeepers are used to group gateways into logical zones and perform call routing between them.
Gateways are responsible for edge routing decisions between the Public Switched Telephone Network
(PSTN) and the H.323 network. Cisco gatekeepers handle the core call routing among devices in the
H.323 network and provide centralized dial plan administration.
Without a Cisco gatekeeper, explicit IP addresses for each terminating gateway would have to be
configured at the originating gateway and matched to a Voice over IP (VoIP) dial-peer. With a Cisco
gatekeeper, gateways query the gatekeeper when trying to establish VoIP calls with remote VoIP
gateways.
Some of the gatekeeper messages

GRQ/GCF/GRJ (discovery)
Unicast or multicast, find a gatekeeper
RRQ/RCF/RRJ (registration)
Endpoint alias/IP address binding, endpoint authentication
ARQ/ACF/ARJ (admission)
Destination address resolution, call routing
LRQ/LCF/LRJ (location)
Inter-gatekeeper communication
BRQ/BCF/BRJ (bandwidth modifications)
DRQ/DCF/DRJ (disconnect)
Call termination
RAS Gatekeeper Registration
Illustrated
Gatekeeper
RRQ RRQ
Hello: I am Registering My
Hello: I am Registering My
Name or E.164 Address
Name or E.164 Address RCF RCF (Gateway B - Prefix 416)
(GW-A - PREFIX: 514)
IP QoS
GW A WAN GW B
RAS—Registration Admission and Status
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UDP Transport Port 1719
RRQ—Registration Request
RRJ—Registration Reject
RCF—Registration Confirm
109
Address Translation—Translates H.323 IDs (such as gwy1@domain.com) and E.164 numbers

(standard telephone numbers) to endpoint IP addresses.
Each gateway will register to Gatekeeper with an ID known as H323 Alias. Gatekeeper identifies
the gateway using these IDs.
As gateway register to the gatekeeper, gateway may have the capability to advertise all the prefix it
can reach. For example: if Toronto gateway is connected to city of Toronto with area code
416XXXXXXX, gateway may advertise Prefix 416 to Gatekeeper.
Gatekeeper builds a dynamic table as gateway register. In the table, it contains the prefix that
gatekeeper learned as well as the IP address of the gateway.
RAS Call Admission Illustrated
GKA (VOICERACKXX)
Dynamic Table
ARQ (Admission Request): GW-B
I Have a Call for Prefix 416
416-839-1717 IP: 1.1.1.1
GW-A
IP QoS Prefix: 514
ARQ WAN IP: 2.2.2.2
ACF
H.323 Call Set-Up

GW-a GW-B
ACF (Admission Confirm):
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Yes You Can, Use GW-B
IP Address 1.1.1.1
110
• Admission Control—Controls endpoint admission into the H.323 network. In order to achieve
this, the gatekeeper uses these:
•H.225 Registration, Admission, and Status (RAS) messages
•Admission Request (ARQ)
•Admission Confirm (ACF)
•Admission Reject (ARJ)
• Bandwidth Control—Consists of managing endpoint bandwidth requirements. In order to
achieve this, the gatekeeper uses these H.225 RAS messages:
•Bandwidth Request (BRQ)
•Bandwidth Confirm (BCF)
•Bandwidth Reject (BRJ)
• Zone Management—The gatekeeper provides zone management for all registered endpoints in
the zone. For example, controlling the endpoint registration process.
Scaling Gatekeepers: H.323 Zones
H.323 H.323
Gatekeeper A QoS Gatekeeper B
WAN
GK GK
Gatekeeper Gatekeeper
Zone A Zone B
H.323
GW
Local Local Local

PSTN PSTN PSTN
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111
Gatekeeper can used to scale network to very large.
Gateway typically register to a zone within a gatekeeper. That zone is consider as a local zone.
When a zone belongs to another gatekeeper, that zone is consider as a remote zone.
For example: ZoneA is a local zone to gatekeeper A while Zone B is a local zone to gatekeeper B.
ZoneA is a remote zone to gatekeeper.
Gatekeeper Inter-zone
Communication:
GKA GKB
LRQ
LCF
ACF ACF
IP Network
ARQ H.225 Fast Start ARQ

H.225 Fast Connect
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UK
Gateway A Gateway B
Phone A 3001 Phone B 5001
112
In Inter-zone, Gatekeeper to gatekeeper, LQR messages are sent. LRQ stands for Location Request
Query. Gatekeeper to Gatekeeper configuration must be manually defined. On GKA, you must define a
remote gatekeeper which happens to be a local zone of GKB and vice versa. Then you must use zone
prefix to route calls to other gatekeeper. Gatekeeper do not exchange any information with each other.
Gatekeeper Scaling: Directory Gatekeeper
Small Network—Gateways Only Small Network—Simplified with a Gatekeeper
VoiceBootcamp 113
As you can see in a small network of 8 gateways if you were to deploy a fully mash, number of
dial peer that you will have to create may become an administrative nightmare.
114
Cisco IOS Gatekeeper: Common

Terms
• Zone—a collection of nodes for routing calls

(can be H.323 clients, CallManager clusters, or H.323
Gateways)
• Zone Prefix—a string of numbers used associate a dialed
number to a zone
• Tech Prefix—a unique string used to group endpoints of
the same type together
• Default Technology—Gateways that register with a tech
prefix that are used for default routing of any E.164
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address that is otherwise unresolved
114
Zone Prefixes
• A zone prefix is the part of the called number that identifies the zone to which a call hops off.
Zone prefixes are usually used to associate an area code to a configured zone.
• The Cisco gatekeeper determines if a call is routed to a remote zone or handled locally. For
example, according to this configuration excerpt, gatekeeper (GK) A forwards 416....... calls to
GK-B. Calls to area code (408) are handled locally.
Technology Prefixes
• A technology prefix is an optional H.323 standard-based feature, supported by Cisco gateways
and gatekeepers, that enables more flexibility in call routing within an H.323 VoIP network. The
Cisco gatekeeper uses technology prefixes to group endpoints of the same type together.
Technology prefixes can also be used to identify a type, class, or pool of gateways.
• Think of tech prefix is like a TAG. Based on that TAG you can route calls to the gateway that
own that tag.
• Default Technology prefix is a gateway of last resort.
Gatekeeper Call Routing: Zone Prefixes &
Default Technology Prefixes
1 GK = Gatekeeper
Call to
14163133001 10
11
Local Zone: SFO 2 Local Zone: TOR
Zone Prefix: 1408* Zone Prefix: 1416*
GK
SFO Phone 1 SFORTR Default Technology Prefix: 1# TORRTR TOR Ph-1
14086165001 Technology Technology 14163133001
Prefix: 1# Prefix: 1#
1 ARQ to 14163133001 14163133001 registered? No 6
2 Technolgy prefix match? No Was a technology prefix found? No 7
3 Zone prefix match? Yes Default technology prefix set? 1# 8
Select a gateway in TOR with

4 Target zone = TOR 9
VoiceBootcamp
technology prefix 1#.
5 Is TOR a local zone? Yes ACF, destination TORRTR 10
115
As the call arrive to gatekeeper, gatekeeper first look at the number and try to match a technology
prefix. Now if technology prefix is found then next step is to match against zone prefix. However in
order to generate either ACF or LRQ, Gatekeeper has to determine if the zone prefix is local or remote.
If it is remote then it will generate LRQ message accordingly. Otherwise it will response with ACF if
permitted. Now zone the zone is matched, it will try to figure out if it is local or remote. If it is local
then next task is to see if the number that user dialed is actually registered in Gatekeeper. Often
Gateway does register E.164 number. If number is not registered and technology prefix was not found
then gatekeeper will try to use default technology prefix if configured. Otherwise call will fail.
Technology Prefixes
1 GK = Gatekeeper
Call to
14163133001 10
11
GK
SFO
SFO Phone 1 TOR TOR Ph1
Technology Prefix: 1#
14086165001 Technology 14163133001
Dial Peer Technology
Prefix: 1#
Prefix: 1#
1 ARQ to 1#14163133001 Is TOR local? Yes 6
2 Technolgy prefix match? Yes, 1# 14163133001 registered? No 7
3 Hopoff prefix? No Was a technology prefix found? Yes 8
Select a gateway in TOR with tech

4 Zone prefix match? Yes prefix 1#.
9
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5 Target zone = TOR ACF, destination TOR 10
116
As the call arrive to gatekeeper, gatekeeper first look at the number and try to match a technology prefix.
Now if technology prefix is found then next step is to match against zone prefix. However in order to
generate either ACF or LRQ, Gatekeeper has to determine if the zone prefix is local or remote. If it is
remote then it will generate LRQ message accordingly. Otherwise it will response with ACF if permitted.
Now zone the zone is matched, it will try to figure out if it is local or remote. If it is local then next task
is to see if the number that user dialed is actually registered in Gatekeeper. Often Gateway does register
E.164 number. If number is not registered and technology prefix was found then gatekeeper will use
select a gateway with that tech-prefix.
Registered Numbers
1 GK = Gatekeeper
Call to
14163133001 7
8
GK
SFO Phone SFORTR TORRTR TOR Ph 1
14086165001 Technology Technology Prefix: 1# 14163133001
Prefix: 1# E.164 14163133001
1 ARQ to 14163133001 14163133001registered? Yes 6
2 Technology prefix match? No ACF, destination TORRTR 7
3 Zone prefix match? Yes
4 Target zone = TOR
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5 Is TOR a local zone? Yes
117
As the call arrive to gatekeeper, gatekeeper first look at the number and try to match a technology
prefix. Now if technology prefix is found then next step is to match against zone prefix. However in
order to generate either ACF or LRQ, Gatekeeper has to determine if the zone prefix is local or
remote. If it is remote then it will generate LRQ message accordingly. Otherwise it will response
with ACF if permitted. Now zone the zone is matched, it will try to figure out if it is local or remote.
If it is local then next task is to see if the number that user dialed is actually registered in
Gatekeeper. Often Gateway does register E.164 number. If number is registered then gatekeeper
will simply reply with ACF message and permit the call.
Cisco IOS GK Configuration Basics
gatekeeper
zone local <zone_name> <domain>
zone remote <zone-name> <domain> <ip_addr>
zone prefix <zone_name> <E.164 string>
gw-type-prefix <E.164 string> <option>
bandwidth <interzone | remote | session | total> <kbps>
VoiceBootcamp 118
• <zone_name>—the logical name of the zone (ie. TOR, SFO, UK, etc…)
• <domain>—domain of the zone (ie. inecanada.com, gk.voicebootcamp.com)
• <E.164 string>—the prefix that a given zone will handle (416*, 514*, 408*)
• <option>—other options to further influence call routing (ie. default-technology, static GW and
zone hopoff)
• <kbps>—the amount of bandwidth to allow within and between zones (G711 = 128kbps, G729 =
16kbps)
Cisco IOS GK Configuration Example
gatekeeper
zone local VOICERACKXX voicebootcamp.com XX.11.11.11
zone remote BACKBONE voicebootcamp.com 135.11.11.11 1719
zone prefix BACKBONE 011*
zone prefix VOICERACKXX 3... gw-priority 10 trunk_2
zone prefix VOICERACKXX 3... gw-priority 0 UKGW
zone prefix VOICERACKXX 44* gw-priority 0 trunk_2
zone prefix VOICERACKXX 44* gw-priority 0 trunk_1
zone prefix VOICERACKXX 44* gw-priority 10 UKGW
no shutdown
VoiceBootcamp 119
For Basic Gatekeeper configuration, you have to first enter in to gatekeeper config mode.
To define local zone type the following command. Local zones are used to manage gateways.
Gateway can only be part of one local zone. When defining a local zone, domain name does not
have to be a valid one. Although IP address is not mandatory but it is recommended that you
define a loopback address
zone local VOICERACKXX voicebootcamp.com XX.11.11.11
Remote zones are zone that are managed by other gatekeeper. Remote zone do not register with
gatekeeper. They simply point to another gatekeeper via IP address. All zone names are case
Sensitive
zone remote BACKBONE voicebootcamp.com 135.11.11.11 1719
Gatekeeper use zone prefix command after tech-prefix to decide where the call should to go. Here
for example I am stating that any calls with 011 should be routed to backbone gatekeeper.
zone prefix BACKBONE 011*
Following two commands are used to route call starting with 3… to CallManager. When
callmanager register to the gatekeeper, it changes its trunk name and add an increment value of 1
to each server. For example for publisher it will name the trunk as TRUNK_1, for subscriber it will
be named as TRUNK_2 so and so.

Disable zone-prefix for specific gateway, define priority of zero. In this example, any call starting
with 3… should not be sent to UK Gateway.
Enable gatekeeper
no shutdown
VoiceBootcamp
H323 GATEWAY
• Interface loopback 0
Ip address XX.33.33.33 255.255.255.255 Enable H323 on this Interface
H323-gateway voip interface Define a h323 alias
H323-gateway voip h323-id UKGW Register to GK XX.11.11.11 with zone

RACKXX
H323-gateway voip id VOICERACKXX ipaddr XX.11.11.11
H323-gateway voip tech-prefix 1# Disable gateway functions
Enable Gateway to register to GK
• No gateway
• Gateway
• Dial-peer voice 3000 voip Dialpeer sends call to RAS which is
gatekeeper
Destination-pattern [3,5]…
Session target ras
VoiceBootcamp
When use rdial 3001 or 5001 this dial peer
will add 1#. So gatekeeper sees the
Tech-prefix 1# incoming call as 1#3001
Based on the 1# and zone prefix, GK will
route the call accordingly.
Num-exp should be used to strip the

Tech-prefix 120
H323 Gateway have atleast one interface with h323-gateway settings. Source address of H323 traffic
must be configured properly otherwise CallManager may not route calls properly
Once interface level is configured, on a global configuration you must type no gateway and gateway to
activate the registration
If gateway is receiving traffic with tech-prefix ensure that translation rule or num-exp is used to remove
the tech-prefix.
Outbound dial peer must use session target ras instead of IP address, gateway already knows which
gatekeeper to send the traffic to.
Call Manager Configurations for Gatekeeper
VoiceBootcamp 121
To route calls via Gatekeeper, you must add gatekeeper and trunk.
Under Device Menu, go to Gatekeeper and add a gatekeeper reference. Once gatekeeper reference
is added, Trunk must be configured which allows you to join a particular zone in a gatekeeper.
Call Manager Configurations for Gatekeeper
(cont’d)
VoiceBootcamp
Translation pattern must be used
To remove tech-prefix
122
Under Gatekeeper Information
Make sure you select the zone name (case sensitive) and tech-prefix if required.
Cisco IOS GK Verification
Commands
GK#show gatekeeper ?
calls Display current gatekeeper call status
circuits Display current gatekeeper circuits
clusters Display gatekeeper cluster info
endpoints Display all endpoints registered with this gatekeeper
gw-type-prefix Display Gateway Technology Prefix Table
performance Display gatekeeper performance data
servers Display gatekeeper servers info
status Display current gatekeeper status
zone Display zone information
GK#show gatekeeper zone prefix

ZONE PREFIX TABLE
=================
GK-NAME E164-PREFIX
------- -----------
TOR 1416*
SFO 1408*
UK 4402*
VoiceBootcamp 123
show gatekeeper calls - Display current gatekeeper call status

show gatekeeper circuits Display current gatekeeper circuits
show gatekeeper clusters Display gatekeeper cluster info
show gatekeeper endpoints Display all endpoints registered with this gatekeeper
show gatekeeper gw-type-prefix Display Gateway Technology Prefix Table
show gatekeeper performance Display gatekeeper performance data
show gatekeeper servers Display gatekeeper servers info
show gatekeeper status Display current gatekeeper status
show gatekeeper zone Display zone information
Gatekeeper Debug
Debug gatekeeper main 10 or 5

hidden command that allows you to see gatekeeper
activity.
VoiceBootcamp 124
Debug gatekeeper main 10 or 5 is a hidden command and provide detail information such as why the call
failed?
Advanced Gatekeeper
Zone Prefix
gatekeeper
zone prefix VOICERACK66 3… gw-priority 10 trunk_2
Default Technology Prefix
gatekeeper
gw-type-prefix 1#* default-technology
Backbone Gatekeeper
Gatekeeper
zone remote BACKBONE inecanada.com 135.11.11.11 1719
Zone Security
VoiceBootcamp
Gatekepeer
no zone subnet VOICERACK66 default enable

zone subnet VOICERACK66 135.XX.100.11 /32 enable  host based enable
125
Zone Prefix – Is used to define static prefix and endpoint that are responsible for this prefix
Default Technology Prefix – When Gatekeeper receives call with a tech-prefix or a number that it does
not know what to do with since there is no explicit configuration for it, it will route the call to a gateway that
has registered to the gatekeeper with a tech-prefix marked as Default Technology Prefix
Remote Zone – Remote zone are zones that are managed and configured on another gatekeeper.
Zone Security – By default any h323 gateway that knows the IP address and zone name of the
gatekeeper will be able register. Using Zone subnet, you can disable and enable which gateway can
register based on their source IP address. However first you must disable all gateway and then enable
explicitly one by one.
Exam Tips: Make sure you configure basic gatekeeper and ensure all h323 gateway can register.
Then block their registration. In case if you put too many configurations, you may not know what
the problem.
Adv. Gatekeeper Contd
• Bandwidth
Gatekeeper
bandwidth total default 512 Specifies the default value for all zones
bandwidth total zone VOICERACK66 512  Specifies the total amount of bandwidth for H.323 traffic
allowed in the zone
bandwidth intrazone VOICERACK66 64  Specifies the total amount of bandwidth for H.323 traffic from
the zone to any other zone.
bandwidth session zone VOICERACK66 16  Specifies the maximum bandwidth allowed for a session in the
zone.
VoiceBootcamp
126
The Cisco Gatekeeper can reject calls from a terminal due to bandwidth limitations. This can occur if the
Gatekeeper determines that there is not sufficient bandwidth available on the network in order to support
the call. This function also operates during an active call when a terminal requests additional bandwidth
or reports a change in bandwidth used for the call.
The Cisco Gatekeeper maintains a record of all active calls so that it can manage the bandwidth
resources in its zone
When you decide whether there is enough bandwidth in order to accept a call Admission Request (ARQ),
the Cisco Gatekeeper calculates the available bandwidth with this formula: Available_bandwidth =
(total_allocated_bandwidth) - (bandwidth_used_locally) - (bandwidth_used_by_all_alternates).
If the available bandwidth is sufficient for the call, an Admission Confirmation (ACF) is returned, otherwise
an Admission Rejection (ARJ) is returned
Dial-plan Considerations
VoiceBootcamp 127
The dial plan is the most fundamental attribute of a telephony system. It is at the very core of the user
Experience because it defines the rules that govern how a user reaches any destination. These rules
include
Dial Plan
The “IP Routing” of IP Telephony
Route Head office
Pattern 9.1416XXXXXXX
Gatekeeper
Cluster
CallManager
GK
5000 IP WAN
Router/GW PSTN
3001 914163133001
CallManager Routes Two Basic Call Types:
VoiceBootcamp
• On-Net Calls—Destination Directory Number (DN)
is registered with CallManager
• Off-Net Calls —External route patterns must be
configured on CallManager
128
Call Classification can be changed at the gateway levels or at Route Pattern. Calls that originate and
terminate on the same telephony network are considered to be on-network (or on-net). By contrast, if a
call originates in company A and terminates at company B, it probably has to be routed through different
telephony networks: first company A's network, followed by the PSTN, and finally into company B's
network. From the caller's perspective, the call was routed off-network (or off-net); from the called party's
perspective, the call originated off-net.
Dial Plan
CallManager Call Routing Logic
CallManager Call Routing Logic
Route Patterns
User Dials
“6500” 6XXX
62XX
User Dials Directory Numbers
“6234” 6234
6234
• CallManager matches the most specific pattern (longest-
VoiceBootcamp
match logic)
• An IP phone directory number is a special case
of route pattern that matches a single number
129
Longest prefix match will always be selected first. However if there is a DN that matched the dialled
number that that DN will be matched
Defining External Routes
External Route Elements in CallManager
Route Pattern Route

• Matches dialed number for external calls Pattern
• Performs digit manipulation (optional)
•In IOS we call this dial-peer
Configuration Order
Route
Route List List
• Chooses path for call routing 2nd
1st Choice
• Points to prioritized route groups Choice
Route Route
Group Group
Route Group 1st 2nd
• Choose the right devices. Choice Choice
GK
IP WAN 6608 t1
Devices &
1 fxo
• Gateways (H.323, MGCP)
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• Gatekeeper
• Inter-Cluster Trunk (remote CM)
130
Route Patterns
Route patterns are strings of digits and wildcards, such as 9.[2-9]XXXXXX, configured in
Cisco Unified CallManager to route calls to external entities. The route pattern can point
directly to a gateway for routing calls or point to a route list, which in turn points to a route
group and finally to a gateway.
Route Lists
A route list is a prioritized list of eligible paths (route groups) for an outbound call. Typically, a
route list is associated with a remote location, and multiple route patterns may point to it. A
typical use of a route list is to specify two paths for a remote destination, where the first
choice path is across the IP WAN and the second-choice path is through the local PSTN
gateways.
Route Groups
Route groups control and point to specific devices, which are typically gateways (MGCP or
H.323), H.323 trunks to a gatekeeper or remote Cisco Unified CallManager cluster, or SIP
trunks to a SIP proxy. (In Cisco CallManager Release 3.2 and earlier, the role of the H.323
trunk was performed by the Anonymous Device gateway and by H.323 gateways configured
using the Intercluster Trunk protocol.)
Route Group
VoiceBootcamp
131
Route group
Route Group is used to decide which gateway to hand the call over to. Usually a route group contains
gateway from single site. For example: GW1 belongs to Toronto, Canada while GW2 belongs to New
York. Now you do not want to put both GW1 and GW1 in the same route group since they represent two
different area therefore numbering can conflict. If you add a 2nd gateway in Toronto for backup such as
GW3 than add GW3 and GW1 in to single route group with GW1 being the top priority.
Route List
VoiceBootcamp To use Local Route group

Route List must contain this RG
133
Route List
Route List is used to decide which path to use to route calls to. For example you may have one route
group for PSTN while 2nd Route Group for IP WAN. Now to save long distance bill you want to use 2nd
Route Group as a first choice while 1st route group as a 2nd choice. In order to achieve this you must put
both of these gateway to the Route List and list the 2nd one at the top.
Local Route Group
Device must have local route group selected for

This feature to work.
VoiceBootcamp 134
Route Pattern
VoiceBootcamp Select the Route List from the list
135
• The Local Route Group feature helps reduce the complexity and maintenance efforts of provisioning in
a centralized Cisco Unified Communications Manager deployment that uses a large number of
locations. The fundamental breakthrough in the Local Route Group feature comprises decoupling the
location of a PSTN gateway from the route patterns that are used to access the gateway
• Use of Local Route group can reduce number of router pattern, route list and route group requirement
Partition and CSS
EVERYONE CAN REACH EACH OTHER
VoiceBootcamp
136
• A partition is a group of directory numbers (DNs) with similar accessibility, and a calling search
space defines which partitions are accessible to a particular device. A device can call only those
DNs located in the partitions that are part of its calling search space.
• Items that can be placed in partitions all have a dial able pattern, and they include phone lines,
route patterns, translation patterns, CTI route group lines, CTI port lines, voicemail ports, and
Meet-Me conference numbers. Conversely, items that have a calling search space are all
devices capable of dialing a call, such as phones, phone lines, gateways, and
Partition and CSS
Phone C
Phone A
Partition = A Partition = C
NO ONE CAN REACH EACH OTHER
Partition = A Phone D
VoiceBootcamp
Phone B
Parition blocks inbound communication unless calling party has CSS with
Called Party’s partition in it
Partition = B
137
When IP phone belongs to a partition, all incoming calls to that IP Phone automatically gets blocked
Unless calling party has the necessary permission to call this partition.
Two phones in the same partition alone does not mean they can talk to each other. You will still need
CSS for each phone to talk to each other.
138
CSS
Partition = C CSS_C
Partition = A Partition C
A Partition B
CSS_A Phone A B, D but not C C

Partition A Phone B A, C but not D
Partition B
Phone C D only
Phone D C, A and B
VoiceBootcamp
B
CSS_B CSS_D
Partition A Partition C
Partition = A Partition = B
Partition C Partition A
138
In this example, we have created a Calling Search Space for each phone. As you can see from
the arrange that Phone B is able to dial A, C while not D.
Partition Example
Go to: Route Plan  Class
of Service  Partition or
Calling Search Space
VoiceBootcamp 139
When creating partition and Calling Search, ensure that proper naming is followed.
NOTE: DO NOT ASSIGN PARTITION TO PHONE UNLESS SPECIFIED. ASSIGNING

PARTITION TO PHONE CAN CAUSE ISSUES LATER IN THE EXAM.
Partition and CSS Example
Local calls
DD - Pre-dot
Prefix – 1408
CSS_TOR_LOCAL
PT_TOR_911
TOR – 6608 T1
PT_TOR_LOCAL
RG_TOR
TOR TOR-A 8391717
9.[2-9]XXXXXX
RL_TOR_LOCAL
PT_TOR_LOCAL DD - Pre-dot
TOR-S Prefix – N/A
9.8391717
9.6391717
RL_SFO_LOCAL
9.[2-9]XXXXXX 8391717
RG_SFO
SFO PT_SFO_LOCAL
CSS_SFO_LOCAL SFO MGCP
PT_SFO_911
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PT_SFO_LOCAL DD - Pre-dot
Prefix – N/A => Prefix – 1416
140
In this example, when user dials 98391717, it will match the pattern 9.[2-9]xxxxxx which is pointed to
RL_TOR_LOCAL. Now RL_TOR_LOCAL has two route group. First one is RG_TOR while 2nd back up
is RG_SFO. When call arrives in RL_TOR_LOCAL it will go to RG_TOR where 9 will be removed due to
Pre-Dot. Call will out as 8391919 and PSTN will route it to the correct phone.
Now if RG_TOR is not available because 6608 is down or something, then call will be routed to RG_SFO.
However, pre-dot will remove 9 so the call by default will go out as 8391717. This can be troublesome as
it might end up ringing a phone in San Francisco. Therefore in order to re-route this call to Toronto, you
must add 1416 as a prefix. So 8391717 become 14168391717.
141

LD – To North America
CSS_TOR_LOCAL
TOR – 6608 T1
PT_TOR_911
PT_TOR_LOCAL
RG_TOR
PT_TOR_LD
DD - Pre-dot TOR
Prefix – N/A
9.1[2-9]XX[2-9]XXXXXX TOR-A
PT_TOR_LD
RL_TOR_LD
9.14088391717 DD - Pre-dot
Prefix – N/A
TOR-S
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RG_SFO SFO
SFO MGCP
Since it is a long distance call, there is no prefix require. LD

Calls from any where in North America is same as of Jun 18 5007
141
In this example, when user dials 914088391717, it will match the pattern 9.1[2-9]xx[2-9]xxxxxx which is
pointed to RL_TOR_LD. Now RL_TOR_LD has two route group. First one is RG_TOR while 2nd back up
is RG_SFO. When call arrives in RL_TOR_LOCAL it will go to RG_TOR where 9 will be removed due to
Pre-Dot. Call will out as 14088391919 and PSTN will route it to the correct phone.
Now if RG_TOR is not available because 6608 is down or something, then call will be routed to
RG_SFO. However, pre-dot will remove 9 so the call by default will go out as 14088391717.
LD calls from TOR to SFO only use SFO GW
14088391717
CSS_TOR_LOCAL
TOR – 6608 T1
PT_TOR_911
PT_TOR_LOCAL
RG_TOR
PT_TOR_LD
PT_TOR_LD_SFO
DD - Pre-dot TOR
Prefix – 1408
9.1408[2-9]XXXXXX
9.1[2-9]XX[2-9]XXXXXX
91408.[2-9]XXXXXX TOR-S
PT_TOR_SFO_LD
RL_TOR_LD_SFO
9.14088391717 DD - Pre-dot
Prefix – N/A
TOR-A
RG_SFO SFO
SFO MGCP
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8391717
Since TOR LD calls use 6608 T1 as a first gw, and SFO as a 2nd. This
Task requires you to route calls to SFO first and then 6608 only if
Toronto calls SFO area code 1408.
142
In this requirement, long distance calls from Toronto to SFO must take SFO GW. Typically Toronto
LD calls are routed via 6608 which is the first priority. However when area code 1408 is dialed, it
must be routed to SFO Gateway.
RL_TOR_LD_SFO we must have SFO Gateway as a first priority. However, Pattern must also be
Change since LD pattern is generic. So we need more specific pattern matching 1408 and pointed to
a new Route List such as RL_TOR_LD_SFO which has SFO Gateway first. However, keep in mind
That when call is routed via Toronto gateway call must be routed as 11 digits.
TOR to UK using 4 digit dialing with Access Code
Any calls from Toronto to UK should use Gatekeeper and then 6608
As a backup
CSS_TOR_LOCAL
Gatekeeper
PT_TOR_911
PT_TOR_LOCAL
RG_GK
PT_TOR_LD
PT_TOR_TOLL
DD - Pre-dot
IP WAN
Prefix – 1#
8.6XXX TOR-A
PT_TOR_TOLL
RL_TOR_TOLL
86001 TOR-S DD - Pre-dot
PSTN UK
Prefix – 0114402896
RG_TOR
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TOR 6608 T1
UK gateway is registered with 1# to the gatekeeper as a tech-prefix.

Therefore any calls to UK must include 1# at the beginning of the
Number.
143
In this example, when user dials 86001 it will match the pattern 8.6XXX which is pointed to
RL_TOR_TOLL. Now RL_TOR_TOLL has two route group. First one is RG_GK while 2nd back up is
RG_SFO. When call arrives in RL_TOR_TOLL it will go to RG_GK where 8 will be removed due to Pre-
Dot. Call will out as 1#6001. We need to add 1# since gatekeeper is expecting 1 # as a tech-prefix.
Now if RG_GK is not available then call will be routed to RG_TOR. You must prefix 011440289X since it
is an international call.
Local Route Group
DD - Pre-dot
Device Pool 5 Prefix – 1408
CSS_TOR_LOCAL RG_TOR TOR – 6608 T1
PT_TOR_911
PT_TOR_LOCAL
RG_TOR
TOR 8391717
9.[2-9]XXXXXX 4
PT_TOR_LOCAL DD - Pre-dot
Prefix – N/A
9.8391717 1
RL_PSTN Local Route Group
9.6391717 2 3
9.[2-9]XXXXXX 8391717
RG_SFO
SFO PT_SFO_LOCAL
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CSS_SFO_LOCAL SFO MGCP
PT_SFO_911
PT_SFO_LOCAL Device Pool DD - Pre-dot
Prefix – N/A => Prefix – 1416
RG_SFO
144
In this example, when end user dials 93013001, call will hit the route-pattern 9.[2-9]XXXXXX, then it is
transferred to RL_PSTN. RL_PSTN send the calls to special route group call Local Route Group which
tells the CallManager to use the originating device’s (calling party) device pool route group setting.
CallManager looks at the device pool of Toronto IP Phone and realize that it has a Route Group call
RG_TOR. So the call goes to Toronto Route Group.
Configuring External Phone Number
Mask
–Go to Device > Phone > Find

and select the corresponding
phone
–Under Association Information,
click the corresponding Line
–Scroll down to Line x on Device
configuration (see picture)
–Type full E.164 PSTN number
in the External Phone Number
Mask field
–In the Route Patterns that point
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to PSTN (e.g. 9.! or 9.@), scroll
to Calling Party
Transformations
–Check the Use Calling Party's
External Phone Number Mask
145
option
External Phone Number mask can be configured on Phone level or defined during auto-registration
In order for CallManager to replace the caller ID with external Phone number must, on the route pattern or
in the route list you must select Use calling party’s Phone’s external phone number mask
Digit Prefix
– Add digits to the pattern

–Valid entries include the digits 0 through 9, *, and #
–Part of Calling/Called Transformations settings
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146
• Digit Prefix can be configured on both Calling Number as well as Called Number
Calling Party Transformation Order
1. Apply the external Directory Number 45062

phone number mask
External Phone
2. Apply the calling party 41671XXXXX
Number Mask
transformation mask
4167145062
3. Apply prefix digits Calling-Party
Transformation 51485XX000
Mask
Caller ID 5148545000
√
41685XX000
VoiceBootcamp 147
The calling party number associated with a call routed through Unified CM might sometimes have to be adapted
before it is presented to a phone or to the PSTN. Calls offered to gateways might require the calling party number
be manipulated to adapt it to the requirements of the telephony carrier to which the gateway is connected. For
example, a call from +1 416 725 4000 offered to a gateway located in France might have to represent the calling
number as 00 1 416 725 4000, with a Calling Party Number Type set to International.
Toll Fraud Control
•Toll Fraud is a feature that can be mis-used by both internal as well

as external users
• A typical toll fraud when a en external user pretend to be a an

employee calls the receiptionist and request her to transfer the call to
another country because he/she was suppose to join a meeting. This
is an external threat
•Internal threat is when an employee decide to forward all calls to

another country to one of their relative, so when he/she calls at night
call will automatically be forwarded to an international number
• Toll Fraud can be control using Transfer and call forward restriction.
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148
Call Classification
Calls using this route pattern can be classified as on‐net or off‐net calls. This route pattern can be used to prevent
toll fraud by prohibiting off‐net to off‐net call transfers or by tearing down a conference bridge when no on‐net
parties are present.
When the "Allow device override" box is enabled, the calls are classified based on the call classification settings on
the associated gateway or trunk. For example: if Pattern you have Call Classification ON‐NET and Gateway you
have Call Classification OFF NET, result of that call will be classified as: OFF NET if allow device override is checked.

Toll Fraud in CallManager
Toll fraud
– Ability to drop an ad-hoc conference when the conference originator hangs up

– Ability to drop an ad-hoc conference when all internal callers hang up
– Ability to block transfers from external trunks or gateways to external trunks
or gateways
Service Parameters
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149
• In Service parameters under Clusterwide Parameters (Feature - General) configure the Block OffNet To
OffNet Transfer as per requirement.
Toll Fraud in CME
transfer-pattern ….  This basically restrict user to transfer to 4 digit number
Call-forward pattern ….  This basically restrict user to forward to 4 digit number
call-forward max-length 4  This prevent particular DN from being forwarded to a

number that is not 4 digit.
Telephoney-service  after-hours can be used to prevent toll fraud

after-hours
COR List  COR list can be used to prevent unauthorized user from dialing PSTN
Direct-inward-Dial this be used to prevent user from receiving secondary dial tone
thus effectively giving an option to dial out again.
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Transfer Pattern must be configured in order to ensure that IP Phone cannot transfer Off-Net call back to
off-Net. So if you limit the transfer pattern to 4 digit that means IP Phone can only transfer internally.
Media Resources
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151
A media resource is a software‐based or hardware‐based entity that performs media processing functions on the
data streams to which it is connected. Media processing functions include mixing multiple streams to create one
output stream (conferencing), passing the stream from one connection to another (media termination point),
converting the data stream from one compression type to another (transcoding), echo cancellation, signaling,
termination of a voice stream from a TDM circuit (coding/decoding), packetization of a stream, streaming audio
(annunciation), and so forth
Different Types of Media Resources
Media Termination Point Music On Hold

Conferencing
MTP x1000 x6000
V
H.323v1 MOH Server
Transcoding
Call Manager
UCCX
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IP WAN
TOR GW SFOGW
6608
152
Media Termination Point

•Define Supplementary Services.
•Used when SCCP communicate to SIP Devices
•H323 v1 communicate with v2.
Music On Hold
•Provide music when call is on hold
Conference
•Provide resources when party initiate a conference sessions
Transcoding
•Converts codec from G.711  G.729
Software Resources
• Cisco IP Voice Media Streaming Application

MTP (G.711) – one for each server pub and sub
CFB (G.711) – One for each server.
• MoH server
MoH (G.711a, G.711u, G.729) – one for each
server
VoiceBootcamp 153
A software unicast conference bridge is a standard conference mixer that is capable of mixing G.711
audio streams and Cisco Wideband audio streams. The number of conferences that can be supported
on a given configuration depends on the server where the conference bridge software is running and on
what other functionality has been enabled for the application. A media termination point (MTP) is an
entity that accepts two full-duplex G.711 streams. It bridges the media streams together and allows
them to be set up and torn down independently. The streaming data received from the input stream on
one connection is passed to the output stream on the other connection, and vice versa. MTPs have
many possible uses
A software MTP is a device that is implemented by installing the Cisco IP Voice Media Streaming
Application on a server. When the installed application is configured as an MTP application, it registers
with a Cisco Unified CallManager node and informs Cisco Unified CallManager of how many MTP
resources it supports. A software MTP device supports only G.711 streams
Music on hold (MoH) is an integral feature of the Cisco Unified Communications system. This feature
provides music to callers when their call is placed on hold, transferred, parked, or added to an ad-hoc
conference. Implementing MoH is relatively simple but requires a basic understanding of unicast and
multicast traffic, MoH call flows, configuration options, server behavior and requirements
Cisco Unified CallManager allocates and uses the following types of media resources:
•Media termination point (MTP) resources
•Transcoding resources
•Unicast conferencing resources
•Annunciator resources
•Music on hold resources
Conference Bridges
Ad Hoc:
User presses “conf” button; 1st caller put on Hold; gets
dial-tone and dials a second user; presses “conf” again
and all users are now connected on the conference bridge.
Meet-Me:
Conference Controller presses “Meet-Me” button; gets
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dial-tone and dials conf call number; all conf call
attendees calls conference call number.
154
For conferencing, you must determine the total number of concurrent users (or audio streams) required at
any given time. Then you create and configure a device to support the calculated number of streams.
These audio streams can be used for one large conference, or several small conferences. For example, a
conference device that was created with 20 streams would provide for one conference of 20 participants,
or five conferences with four participants each (or any other combination that adds up to 20 total
participants). The total number of conferences supported by each conference device is calculated by
taking the total number of streams (for example, 20) and dividing by three. Therefore, in the example, you
can have twenty divided by three (20/3) or six conferences supported by the conference device.
Although conference devices can be installed on the same PC as the Cisco CallManager, we strongly
recommend against this. If conference devices are installed on the same PC as the Cisco CallManager, it
can adversely affect the performance on the Cisco CallManager.
Conference devices configured for software only support G.711 codecs, however, configuring for
hardware provides transcoding for G.711, G.729 and G.723 codecs.
Media Resources on Cisco IOS
Gateways
–Cisco Unified CallManager media resources can be deployed using

Cisco IOS gateways:
•DSP-based hardware conference bridge
•DSP-based hardware transcoding
•DSP-based hardware Media Termination Points (MTPs)
•Software-based MTPs
–Hardware conference bridges are recommended for remote sites:
•Avoids suboptimal WAN usage for ad hoc conferences
–Transcoding resources usually located in central sites:
•Used to connect G.729 calls to G.711-only applications
VoiceBootcamp
–MTPs connect to media streams using the same codec:
•Can be used to add supplementary services
155
The NM-HDV Farm module ships with two SIMMS and is able to handle three additional SIMMS. Each
SIMM contains three DSPs. Each DSP supports four Transcoding sessions or one Conference Bridge.
Four Transcoding sessions are supported for g729-g711. If you use the Global System for Mobile
communication (GSM), then the DSPs can handle three Transcoding sessions. Therefore, the maximum
number of Transcoding sessions supported by a five-SIMM configuration is sixty Transcoding sessions.
The maximum number of conference calls supported by a five-SIMM configuration is fifteen. The
Conference Bridges and Transcoder sessions configured count against the cumulative total and cannot
exceed the limit of what is supported by the number of DSPs installed
Configuring Conferencing and Transcoding on
Voice Gateway Routers
1. Determine DSP resource requirements

2. Enable SCCP on the Cisco Unified Communications Manager
interface or Cisco Unified Communications Manager Express
3. Configure enhanced conferencing and transcoding
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156
DSP Farm Configuration Example
TOR Montreal
Cisco Unified
Communication IP WAN
s Manager
135.1.100.20
Router1 Router2
sccp ccm 135.Y.100.20

sccp local FastEthernet 0/0
sccp
Phone1-1 Phone1-2 Phone2-1 Phone2-2
3001
voice-card 0
3002 PSTN
dspfarm profile 5 conference
5001 5002
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dsp services dspfarm codec g711ulaw
codec g711alaw
sccp ccm group 22 codec g729ar8
associate ccm 1 priority 1 codec g729abr8
associate profile 5 codec g729r8
register CFGVCBCONF codec g729br8
maximum sessions 1
associate application SCCP 157
A DSP farm is the collection of DSP resources available for conferencing, transcoding, and MTP services. DSP
farms are configured on the voice gateway and managed by Cisco Unified Communications Manager through
Skinny Client Control Protocol (SCCP). The DSP farm can support a combination of transcoding sessions,
MTP sessions, and conferences simultaneously.
Note Hardware MTP services are not supported on the NM-HDV.

Centralized Conferencing Resources
CallManager
X cluster
PSTN
A
IP WAN
B Tor
SFO Conf Site
• Caller X calls A—No voice across WAN

• A conferences B in
WAN VoiceBootcamp
• 3 media/voice streams across
• No conferencing during WAN failures
159
When Conference Bridge is located in Head Office over the WAN, Branch Office IP Phone will use the
CONF Bridge across the WAN when they need such resources. Such design often inefficient due to
more bandwidth utilization during conference services.
Media Resources
Distributed Conferencing Resources
MRGL CallManager MRGL

Cluster
1. SFO 1. TOR
2. TOR
X PSTN
A
IP WAN
B
Device Device
Conf
Pool Conf Pool
MRG=SFO
SFO
MRG=TOR TOR
• Conference between A, B and X—
No voice across WAN
VoiceBootcamp
• Requires extra hardware at branch
MRG = Media Resource Group
• No conferencing during WAN failures MRGL = Media Resource Group List
160
By deploying a local Conference Bridge for Branch office, all media stream will be local when there is a
conference resource. First configure the local router as a Conference Bridge and then added to
Callmanager. Create a media resource group and add this local Conference Bridge. Then create a
Media Resource Group list with this media resource group in it. Then apply the Media Resource Group
list to Device Pool of Branch Office.
Media Resource Group
VoiceBootcamp
Enable Multicasting. Without this, Multicast will not work
Regardless if the router and CCM severs are configured or not
161
You can create separate Media Resource Group for separate resources like
MRG_HW_CONF  This group contains all hardware conferences

MRG_SW_CONF This group contains all software base resources
MRG_HW_XCODER This contains all the transcoder etc.
Now in Media Resource Group, there is no prioritization. If two conference bridge are available in the
group, it may select randomly whichever is available.
If you wish to deploy Multicast Music on Hold, make sure in the Media Resource Group you have atleast
one MOH Server with Multicast enable and the group must have Multicast option check at the bottom.
Media Resource Group List
More then one media resource group can be part of

MRGL. MRGL is a prioritizion using TOP DOWN Approach.
Which ever MRG is at the top will be the active
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Media Resource Group
162
You can create separate Media Resource Group for separate resources like
MRG_HW_CONF  This group contains all hardware conferences

MRG_SW_CONF This group contains all software base resources
MRG_HW_XCODER This contains all the transcoder etc.
Now in Media Resource Group, there is no prioritization. If two conference bridge are available in the
group, it may select randomly whichever is available.
If you wish to deploy Multicast Music on Hold, make sure in the Media Resource Group you have at least
one MOH Server with Multicast enable and the group must have Multicast option check at the bottom.
MoH Overview
The Music On Hold feature provides capability to stream audio to

held users when the MOH feature is enabled.
The MOH server provides Audio Sources and connects a MOH Audio
Source to a number of Streams.
Two types of Hold:

1. User Hold.
2. Network Hold (transfer, conference , call park, etc.)
VoiceBootcamp 163
Music on hold (MoH) is an integral feature of the Cisco Unified Communications system. This feature
provides music to callers when their call is placed on hold, transferred, parked, or added to an ad-hoc
Conference
The basic operation of MoH in a Cisco Unified Communications environment consists of a holder and
a holdee. The holder is the endpoint user or network application placing a call on hold, and the holdee
is the endpoint user or device placed on hold.
Holder decide which music file holdee will listen but holidee decide which server it will receive the
stream from.
MoH multicast server configuration
Uploading New Music File
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164
If multicast is required, then user must enable multicast at every level including audio source that plays
the music.
Enable Multicast on the audio file
Must select
Allow Multicasting
VoiceBootcamp 165
Multicast must be check on the audio file if this file is to be played during multicast sessions. By selecting
multicast does not guarantee that it will work properly unless infrastructure and callmanager is configured
for multicast. There is harm of selecting this even if multicast is not being used.
Must select Enable Multicast

To support Multicast MoH.
Also select increment based
On IP Address. Easier to
Remember
Audio Source File must have

Max-hop set to the appropriate
Values
VoiceBootcamp
166
Ensure that MAX HOPE is set correctly. It is usually over 2.

Add the multicast MoH server in a MRG
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Enable multicast on the group otherwise multicast will not
Work.
167
• Multicast can only be enabled if it is selected in group level.

Multicast on Router
Enable Multicast on each router interface between the IP phone

And CallManager
Ip multicast-routing
Interface serial 0/0

ip pim dense-mode
Interface fastEthernet0/0.101
ip pim dense-mode
no ip igmp snooping
VoiceBootcamp 168
Configure Multicast on every router and interface between the source and the members such as
CCM and IP phones.
Four Levels of Prioritized Audio
• Level four has the highest priority and level one has the
least .
Level four is directory/line based
Level three is device based.
Level two is Device Pool based.
Level one audio source IDs are service wide service
parameters.
VoiceBootcamp 169
• There are four levels of prioritized audio. Level four has the highest priority and level one has the
lowest. The four levels of prioritized audio are described as follows:
• Level four is directory/line based (devices which have no line definition, like gateways, do not have this
level). The system will select the audio source IDs at this level if defined.
• Level three audio source IDs are device based. If none is defined in level four, the system will search
any selected audio source IDs in level three.
• Level two is device pool based. If no level four or level three audio source IDs are selected, the system
selects audio source IDs in level two.
• Level one audio source IDs are service wide service parameters. If levels two, three and four have no
audio source IDs selected, the final level, level one, will be searched for audio source IDs by the
system.
• The held party devices decide which server the audio stream is delivered from. This is based on the
media resource group list (MRGL) configured and where the MRGL is assigned within
Cisco CallManager to the devices.
Media Resource Group Lists
User Needs
Media Resource
Similar to Route Lists Media
Resource
and Route Groups Manager
Assigned to Device
Media Resource
Group List
1st 2nd
Choice Choice
Media Media
Resource Resource
Group Group
1st 2nd 1st 2nd

Choice Choice Choice Choice
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Media Resource Media Resource Media Resource Media Resource
1 2 3 1
170
• Media resource group lists (MRGLs) specify a list of prioritized MRGs. An application can select
required media resources among the available resources according to the priority order defined in the
MRGL. MRGLs, which are associated with devices provide MRG redundancy.
• The preceding figure shows the hierarchical ordering of media resources. It also illustrates that MRGs
and MRGLs are similar to route groups and route lists.
• When a device needs a media resource, it searches its own MRGL first. If none are available, the
device searches the default list. The default list of media resources includes all media resources that
have not been assigned to an MRG. Once a resource is assigned to an MRG, it is removed from the
default list.
MRGL Selection Rules
Two levels of prioritized MRGL selection are implemented.

MRGL at Device level has higher priority.
MRGL at Device Pool level has lower
VoiceBootcamp 171
• There are two levels at which MRGLs can be assigned to devices. The level with the higher
priority is configured at the device level.
• For example, for a phone it is configured at the Phone Configuration page in CallManager
Administration. The lower priority level is an optional parameter of the Device Pool. If a MRGL is
not configured at the device level, it will use the MRGL configured at the device pool level first
and then if there are no resources available, it will try to use resources in the default list.
• If a device has an MRGL configured at the device level, that MRGL is used first and when there
are no resources available from that MRGL, then the device tries to use media resources from
the default list.
Local MoH Source
• Local Cisco Router can be used as a Music source

• Require SRST configurations
Loopback address of the router

• Call-manager-fallback
moh music-on-hold.au
multicast moh 239.1.1.X port 16384 route 5.2.2.2 135.5.65.240
Source Subnet of the

IP Phone
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X depends on what Codec you are using
G.711 is 1, G.729 is 3 etc.
172
Cisco SRST gateways can be configured to multicast Real‐Time Transport Protocol (RTP) packets from
flash memory during fallback and normal Cisco CallManager operation. Cisco CallManager must be
configured for multicast MOH in such a way that the audio packets do not cross the WAN. Audio
packets are broadcast from the flash memory of Cisco SRST gateways to the same multicast MOH IP
address and port number configured for Cisco CallManager multicast MOH.
NOTE: Cisco SRST multicast MOH supports G.711 only

No Multicast across WAN
• If you do not want to allow multicast to cross the WAN in

Branch office
multicast moh 239.1.1.1 port 16384 route 5.2.2.2 135.5.65.240
• Do Not enable multicast on the WAN interface
VoiceBootcamp 173
NOTE: Cisco SRST multicast MOH supports G.711 only

High Availability (SRST & AAR)
VoiceBootcamp 174
Voice Lab Sample Topology
VoiceBootcamp 175
Branch Office IP Phone depends on the head office Call Manager for its functionality. In case of WAN
outage and/or network connectivity problem, BR office IP phone will lose functionality unless SRST is
deployed. SRST provide basic Phone functionality.
176
Survivable Remote Site Telephony (SRST)
Normal Operation CallManager

Cluster
Data Backup Data
Signaling Traffic Traffic
Signaling Traffic
IP WAN
SRST
Router Voice Traffic Central Site
SFO Site
PSTN
Voice Traffic
VoiceBootcamp
• SRST router needs minimal configuration with 3 to 4 lines.
• Remote site IOS router take over SCCP call processing for local ip phones in
case of WAN failure.
• Basic call functions and features are preserved.
176
Cisco SRST provides Cisco CallManager with fallback support for Cisco IP phones that are attached to
a Cisco router on your local network. Cisco SRST enables routers to provide call-handling support for
Cisco IP phones when they lose connection to remote primary, secondary, or tertiary Cisco
CallManager installations or when the WAN connection is down
Cisco CallManager fallback mode telephone service is available only to those Cisco IP phones that are
supported by a Cisco SRST router. Other Cisco IP phones on the network remain out of service until
they reestablish a connection with their primary, secondary, or tertiary Cisco CallManager.
Basic SRST Configurations
Global command to enable SRST Mandatory command to enable router
to receive and process SCCP msgs.
Call-manager-fallback
ip source-address 135.xx.65.240 port 5000 [any-match | strict-match]
max-dn 48
max-ephone 24 Mandatory commands which define the
max-conferences 8 max. # of IP phones and directory
time-format 24 numbers (DNs) supported by SRST.
limit-dn 7960 2 Default is “0”
` dialplan-pattern 1 14086X6... extension-length 4
Limit maximum # strict-match” option

Global prefix which maps full e.164 enables strict ip address
VoiceBootcamp
of DNs assignable
to particular types called number to local ip phone verification of IP phones
of phone. extensions. In this case, if the DID of trying to register to
an inbound call is 14086X65001, it SRST router.
will be routed to a registered DN of
5001. Also being used to construct
full e.164 caller ID for calls
originated from SRST router.
177
At minimum 3 commands are require under SRST configurations
Ip source-address – define which IP address SRST runs
Max-dn - define how many IP phone extensions are allowed. By default 0
Max-ephone – define how many phone to allow to register
**Device Pool in Call Manager decide which phone will have SRST enable and which phone don’t.
SRST MGCP Fallback to H.323
CallManager
Cluster
PSTN
FXO
Gateway
FXS PRI
IP WAN
MGCP Signaling TOR

SFO MGCP H.323 + PRI Backhaul
• Under normal operation, the gateway translates FXS/FXO signaling

into MGCP and backhauls L3 PRI signaling to CallManager
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• When the WAN fails, the gateway reverts to H.323 operation—
SRST provides backup for the IP phones
178
MGCP fallback is a different feature than SRST and, when configured as an individual feature, can be
used by a PSTN gateway. To use SRST as your fallback mode on an MGCP gateway, SRST and MGCP
fallback must both be configured on the same gateway
To make outbound calls while in SRST mode on your MGCP gateway, two fallback commands must be
configured on the MGCP gateway. These two commands allow SRST to assume control over the voice
port and over call processing on the MGCP gateway
SRST MGCP Fallback to H.323 Configuration
ccm-manager fallback-mgcp
Application Allows MGCP gateway to fall

global back to H.323 mode
service alternate DEFAULT
Enables gateway to fall back to
dial-peer voice 1 pots default call application (H.323)
incoming called-number . when mgcpapp is not available
direct-inward-dial
port 1/0:23
!
!
call-manager-fallback
ip source-address 135.XX.65.240 port 5000
max-dn 48
Pots dial-peer for outbound
max-ephone 24 calls in SRST mode. Note that it
dialplan-pattern 1 14086X65... extension-length 4 must have direct-inward-dial,
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otherwise, inbound PRI calls
will get a secondary dial-tone
179
Service alternate Default allows a router to fall back to its default status.
SRST and VOICE Mail
Example 1
call-manager-fallback
ip source-address 135.XX.65.240 port 5000
max-dn 48
max-ephone 24
dialplan-pattern 1 14086X65... extension-length 4
voicemail 914163X33300
call-forward busy 914163X33300
call-forward noan 914163X33300 timeout 10
CallManager 6608 T1 Gateway - Redirecting Number IE Delivery - Outgoing
Step 1 From any page in Cisco CallManager, click Device and Gateway.
Step 2 From the Find and List Gateways page, click Find.
Step 3 From the Find and List Gateways page, choose a device name.
Step 4 From the Gateway Configuration page, check Redirecting Number IE Delivery - Outgoing.
VoiceBootcamp 180
Cisco Unified SRST can send and receive voice-mail messages from Cisco Unity and other voice-mail systems
during Cisco Unified Communications Manager fallback. Calls that reach a busy signal, calls that are
unanswered, and calls made by pressing the message button are forwarded to the voice-mail system
SRST – Call Reoute
• Alias command allows you to re-route the call to alternate

destination
• If ephone-dn and alias has the exact match, by default
ephone-dn is the first priority.
• To change the priority of ephone dn use the following
command under SRST
– max-dn 4 preference 10
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181
Example of Alias command
Alias 1 5… to 50
Any calls to 5XXX

alias 1 5… to 5001 That are not registered
Will be routed to 5001
If call arrive to an extension that is not answering and you

want to re-route that calls to 3001 then use alias with higher
Preference. Ensure that ephone-dn X has a lower preference
value
alias 1 5001 to 5001 cfw 3001 timeout 5
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max-dn 4 preference 2
182
Alias command is used to translate a dialed number in to another number. It only affects DNIS. Now
user dial any number in the range of 5XXX all call will go to 5001 in the 1st example.
2nd Example is call re-routing. Typically what happen when an inbound call arrive to the SRST Router, if
an ephone-dn match occurs, it rings the IP Phone. However by changing the preference in max-dn X
preference 10 you can change the call flow. By doing so, you can have the call hit the alias first before
going to the IP Phone. Not let’s assume our ephone-dn 50 is registered with extension 5001 with a high
preference like 10 (due to max-dn 5 preference 10). You have also configured an alias command as
mention above in the 2nd item. Now if SRST router receives a call for 5001, instead of ringing IP Phone,
it will go to the Alias. Now alias will then forward the call to IP Phone as if alias is calling the phone. Now
since Alias is now handling the call, it is monitoring the call progress. If user does not answer within 5
seconds for example, call will be forwarded to extension 3001.
CFUR
• When IP phone is un-registered in UCM due to network

outage or SRST is activated some how, CallManager can
re-route the call for that IP Phone to their external IP
Phone which will then be re-routed via local gateway to
the SRST router
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183
• Ensure that IP Phone is able to dial Long distance or necessary CSS is applied to phone to make the
call.
Why Call Admission Control?
Example:
WAN bandwidth can only support two calls.
What happens when the third call is attempted?
CallManager CallManager
Call #1
Call #2
X
IP WAN X
Call #3
X
Call #3
Causes poor quality for ALL calls
VoiceBootcamp
Many tools to give voice priority over data.
Call admission control is about preventing voice oversubscription.
184 I
Call Admission Control (CAC) provides mechanisms to control the quantity of calls between two
endpoints. Controlling the number of calls, or the amount of bandwidth that is required between two
endpoints is key to maintaining Quality of Service (QoS) for all existing calls and any new ones. The
network is provisioned to carry a specific amount of Real Time traffic, any traffic exceeding the
provisioned bandwidth, will be subject to delay, jitter and possibly packet loss.
185
Bandwidth used
Bandwidth usedforfor
CAC
CAC
Codec type G.711 G.729
CODEC Type 64Kbps 8Kbps
CallManager Locations 80Kbps 24Kbps
Gatekeeper 128Kbps 16Kbps
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185
The bandwidth figures used for CAC calculations do not take into account sample size, UK headers,
UDP/IP headers or any of the Layer 2 overhead.
This can make a considerable difference in the amount of bandwidth actually used for the call. For
example: When Cisco CallManager requests bandwidth from the gatekeeper during an ARQ or BRQ, It
requests the maximum transmit and receive bandwidth. Therefore for G.711 and G.729, it will use 128k
and 500k respectively. Let us take an example of a gatekeeper configured to admit 256k of bandwidth.
This would allow two calls at G.711. When we factor in the IP, UDP and UK headers, this would be
approximately 80k per call in each direction, a total of 160k. If the same configuration is used and all the
calls are G.729, the gatekeeper will admit 12 calls. With the overhead this would be approximately 24k
per call or a total of 288k in each direction. To maintain our QoS in the WAN we would have to engineer
the links to factor in this variance, resulting in under subscription during the use of G.711 or
heterogeneous use of CODECs. The use of cUK minimizes much of the overhead error, however this is
on a hop-by-hop basis, resulting in each router interface the call traverses having to expand and
compress the UK packet. As the speed of the link and quantity of UK traffic increases, the use of cUK
becomes less desirable.
Centralized Call Processing:
Locations based CAC
Location: SFO
Bandwidth: 256
Applications
(VMail, IVR, ICD, ...) SRST-enabled
PSTN router
CallManager
Cluster
SFO
IP WAN
TOR
Location: None
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Bandwidth: Infinite
UK
Location: SFO
Bandwidth: 96
186
Cisco CallManager provides a simple “Locations” based CAC mechanism for Hub and Spoke Network
Topologies. This is primarily used for Centralized Call Processing. During the configuration of a device on
Cisco CallManager it can be “placed” in a location. The Cisco CallManager has no knowledge of where
the device physically is, if the device moves from one “physical” location to another, without changing the
“location” configuration, Cisco CallManager will incorrectly calculate bandwidth for that device. This will
render the Locations CAC unusable.
As with all Centralized Call Processing deployments, the bandwidth used for a location is not shared
between servers in a cluster. It is therefore important to have only one active server in a cluster. The
other servers in a cluster can be the “Publisher” and or backup server.
187
Location Configuration
VoiceBootcamp
Location is then assigned to devices or Device Pool
187
To place a device in a location, we must first define the locations and the available bandwidth available.
This is achieved from the CCMAdmin pages by going to System>Location. When the locations have
been defined with the available bandwidth, the devices can be configured to be in the location. In the
example above, we defined a location HQ; this has 96Kof available bandwidth that will support up to 4 x
G.729
In the device configuration pages we can specify the location of the device from the drop down menu.
Devices that allow Locations to be defined include phones, gateways and CTI route Points. Phone
devices include IP Phones, CTI Ports and H.323 clients.
The following example shows a gateway defined as HQ that is configured to be in the HQ location. Each
call placed to or from this device, will admitted by Cisco CallManager based on the available bandwidth in
the HQ bandwidth pool. When a call is attempted with insufficient bandwidth available, the call will fail due
to insufficient bandwidth resource and the endpoint will receive a busy tone, additionally IP Phones with a
display will receive a “Not Enough BW” message
Location is applied to device pool or device directly
For non-centralized systems, Cisco Unified Communications Manager offers an alternative CAC method,
Resource Reservation Protocol (RSVP).
With AAR:
• AAR provides a mechanism to automatically reroute

the call through PSTN or other network by using an
external/alternate number when the call is blocked by
Call Manager due to insufficient location bandwidth,
such that the caller does not need to hang up and
redial the called party again.
• In short, AAR is PSTN Backup for Locations.
VoiceBootcamp 188
Withour AAR, call will get fast busy signal when location reject the calls. However in a High
Availability environment that may be unacceptable. AAR can re-route that reject call via PSTN.
AAR Configuration
Service Parameter
AAR must be enable in Service parameter
CSS must have necessary

partition
VoiceBootcamp 189
• To enable the AAR feature for the entire cluster (by default it is enable but double check it)
• AAR Group defines what prefix to add in order to dial PSTN or a Cloude
• AAR Group represents the dialing area where the line/DN, the Cisco voice mail port, or gateway
belongs. AAR Group usually represents different geographical areas (CallManager locations) or area
codes.
• It is assigned to Cisco CallManager Line/DN, Cisco voice mail port, and the gateway device
• The originating DN or device’s AAR Group value, and terminating DN or device’s AAR Group value are
used to index into the AAR Group table to retrieve the prefix digits. For example:
• AAR CSS is required to ensure that if phone is not able to dial certain route-pattern due to CSS
restriction that during AAR, it is allowed. AAR CSS should have enough partition to dial the pattern it
needs to.
AAR Configuration
Service Parameter
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Under Line Level, assign the AAR Group
190
• Each IP Phone or device must have AAR CSS defined on device level
How does AAR work? (cont.)
AAR Instructions Called Number
The call from Phone 1 in TOR to phone A in SFO is 5001

blocked due to Insufficient Bandwidth between
CallManager Locations
CCM retrieves the External/Alternate number for the 4086Y65001

terminating DN (derived from external phone number
mask, e.g. 4086Y6XXXX)
CCM prepends the AAR Group prefix (from Tor to 914086Y65001

SFO, e.g. 91)
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CCM reroute the call using AAR CSS via gateway and
PSTN (assuming there is DID on the terminating side)
914086Y65001
9.1[2-9]XX[2-9]XXXXXX
PT-TOR-LD
191
Here is an example of an AAR call
When TOR Phone 1 dials 5001, location denied the call due to luck of bandwidth. Now CCM
realize that AAR is activated therefore, CCM will look at the database and finds that extension
5001 has an
External phone number mask set to 10 digit. CCM will take that 10 digit and look at the calling
party phone and realize that it belongs to AAR GROUP-TOR while called party phone is in another
AAR Group. Since group decide to add 91 to all call from one group to another, therefore number
becomes 91 follow by 10 digit from external phone number mask. Once it finds a match to a
pattern chances are that this phone may not have access to route pattern due to class of service.
Therefore AAR calling search provide a conditional permission to allow this device to establish AAR call
via long distance or international method.
Unity Connection 7.X
Overview
VoiceBootcamp 192
• CCIE VOICE diagram

• Information Sheet containing DN, IP Address etc
Cisco Unified CM Voice-Mail
Integration
Cisco Cisco
Unified Unity Connection
CM
Cluster
PSTN
–Cisco Unified CM can integrate with Cisco Unity, Cisco Unity Connection, Cisco
Unity Express.
–Cisco Unity and Cisco Unity Connection integrate using SIP or SCCP:
VoiceBootcamp
•SIP integrations include MWI handling.
•SCCP needs additional MWI ports.
–Cisco Unity can handle multiple clusters connected through QSIG tunnels.
–Cisco Unity uses the forwarding information provided by Unified CM to answer
the call appropriately. 193
Cisco Unity Connection supports messaging redundancy and load balancing in an active‐active redundancy model
consisting of two servers, a primary and a secondary, configured as an active/active redundant pair of servers,
where both the primary and secondary servers actively accept calls as well as HTTP and IMAP requests. Both Cisco
Unity and Cisco Unity Connection SIP trunk implementation requires call forking for messaging redundancy
functionality.
Voice-Mail Integration Parameters
Cisco
Unified Cisco
CM Unity Connection
PSTN
Cisco Unified CM parameter Cisco Unity Connection parameter

Number of Voice-Mail Ports Number of Voice-Mail Ports
Message Waiting Information MWI on/off Extension
Voice-Mail Port Name CallManager Device Name Prefix
VoiceBootcamp
Line Directory Number Subscriber Extension
Hunt List, Hunt Pilot, Voice-Mail Pilot,
-
Voicemail Profile
194
One of the important thing in configuring Unity Connection is the Device Name Prefix. If you change it in
the CallManager, make sure exact name is defined in Unity Connection. For example if you change the
name in CallManager to VM then in Unity Connection when you create the Port Group, the name should
be VM-VI (VI is like voice interface)
Unity Connection pulls all the username from CallManager to ensure that Unity Connection is first added
to CallManager as an Application servers.
MWI must match as well.

Voice Mail Integration Elements:
Incoming Call
Cisco Unified CM Incoming Cisco Unity Connection
call
SCCP Directory,
Prompts,
Messages
forwarded
MWI on call
Extension
MWI off
Extension Cisco UnityC
Voice-Mail Message access
Pilot  Hunt Directory access
MWI on Prompt access
Pilot
CM Hunt List call
Voice-Mail SCCP Voice-Mail Port 1
VoiceBootcamp
Port1
Voice-Mail SCCP Signalling Voice-Mail Port 2
Port2 Traffic
Port3
Port4
195
When incoming calls arrive on an IP Phone, under 4 condition call can be routed to voicemail.
Call-Forward All – route to voicemail

Call Forward No Answer – Route to voicemail when no one answer
Call Forward Busy – Route to voicemail when the line is busy
DnD when user press DnD on incoming calls.
This type of call will then hit the Voicemail Profile which is associated with VM Pilot #. VM Pilot number in
return matches the hunt pilot which has a hunt list with line group. Now the line group contains the
voicemail port which is registered by the Unity Connection.
This type of call in Unity Connection is treated as forwarded call.
In Forwarded call, Unity Connection looks

SCCP Voice-Mail Integration
Configuration Procedure
• Cisco Unified Communications Manager SCCP

Integration Tasks:
1. Create MWI extensions
2. Create voice-mail ports
3. Create line group
4. Create hunt list
5. Create hunt pilot
6. Create voice-mail-pilot
7. Create voice-mail-profile
VoiceBootcamp 198
Before logging in to Unity Connection, CallManager must be configured with necessary configurations.
 MWI extension is required so that Unity Connection can advice call manager when to turn the
light on and off.
 Number of voicemail port defines how many simultaneous communication is allowed between
voicemail and callmanager
 Line Group/Hunt-List and Hunt Pilot is required for User to access the voicemail
 Voicemail Pilot Number and Hunt Pilot Number is the same
 Voicemail profile is used by CallManager to assign Voicemail Pilot Number to the message button
of a IP Phone.
Step 1 – Pilot Number & Profile
Select Voicemail Pilot
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Select Voicemail Profile
199
Define Pilot Number and associate with a Voicemail Profile. Multiple Pilot # can be configured.
Step 2 – MWI
Select Message Waiting
VoiceBootcamp 200
MWI – Message Waiting Indicator
Define two number one for ON and one for OFF.

Step 3 – VoiceMail Port
VoiceBootcamp
201
Number of voicemail port will depend on license.

Step 3 – VoiceMail Port (cont’d)
VoiceBootcamp
202
Step 3 – VoiceMail Port (cont’d)
VoiceBootcamp 203
Run the Voicemail port wizard. Ensure that all the settings are define as per the requirement such as
Calling Search Space, Partition etc.
Step 4 – Voicemail Hunt List
VoiceBootcamp 204
Voicemail Hunt list should include the Line group created by the voicemail port wizard application.
Step 5 – Hunt Pilot
VoiceBootcamp
205
Hunt Pilot number and the voicemail pilot number should be the same in most cases.
Step 5 – Call Forward Setting
VoiceBootcamp
206
Every IP phone that has a voicemail mailbox must have its Call forward parameter set properly for each
and every line that has a mailbox.
Step 5 – Adding Unity Conn as a APP
Server
VoiceBootcamp
207
You must add Unity Connection as a Application Server in CallManager otherwise AXL access from Unity
Connection will fail.
Unity Connection Configurations
Select Phone System
Then go to Edit Menu

And Select
Cisco Unified
Communication Manager
AXL server
VoiceBootcamp 208
In unity connection defines the Phone System.

Define UCM and AXL Users
Define the IP address of CallManager and port number is 143

Unity Connection use IMAP port
VoiceBootcamp
Username and password must be the necessary privilege
209
In order for Unity Connection to communicate with CallManager, you must define the AXL Server
settings.
Define Port Group and Ports
Add A New Port Group Port Group is a logical
Group of voicemail port
VoiceBootcamp 210
Port Group and Ports are used to define how much voice mail port will be used. Make sure in Primary
Server Setting you define the IP address of CallManager.
Check Configurations
Go to CallManager  VoiceMail 
VoiceBootcamp
211
You can verify the unity integration by listing the voicemail port from CallManager. Port status should be
registered.
Voicemail Subscriber
VoiceBootcamp
212
Unity Subscribers can be created by either pulling the user from CallManager or from LDAP server
directly.
Note: when importing the users from LDAP, you must define an extension user by users. When importing
users from CallManager, ensure that user has a primary extension defined in their user settings in
Callmanager.
CCME/CUE Configuration: CLI and
GUI
CME
PSTN PSTN-GW
Interfac
e
GUI
CUE initialization wizard
CLI
CME setup
Basic router config
Phones and phone features
Voice gateway config
Extensions
CUE IP addressing
Dial-plans
CUE SIP dial-peers
Vmail setup
Basic CME admin login definition
Mailboxes
CME “Setup” utility
AA setup
Upgrades/Installs
Day-to-day moves, adds
VoiceBootcamp
CUE backup and restore and changes
**Both CLI and GUI Allowed in Lab
213
CLI
• Basic router config

• Voice gateway config
• CUE IP addressing
• CUE SIP dial-peers
• Basic CME admin login definition
• CME “Setup” utility
• Upgrades/Installs
• CUE backup and restore
GUI
• CUE initialization wizard
• CME setup
•Phones and phone features
•Extensions
•Dial-plans
• Vmail setup
•Mailboxes
• AA setup
• Day-to-day moves, adds
and changes
Unity Express - Setup
CME#
!
interface FastEthernet0/0.10X
ip address 135.X.67.240 255.255.255.0
!
interface Service-Engine0/0
ip unnumbered FastEthernet0/0.101
service-module ip address 135.X.67.230 255.255.255.0
service-module ip default-gateway 135.X.67.240
!
ip route 135.X.67.230 255.255.255.255 Service-Engine1/0
!
ip http server
!
VoiceBootcamp
dial-peer voice 6000 voip
destination-pattern 66..
session protocol sipv2
session target ipv4:135.X.67.230
codec g711ulaw
no vad
!
214
When setting up unity express, you must first define the IP UNNUMBER command if you wish to use an
IP address from the same subnet as the main router interface. Then assign the service engine an IP
address.
Static route to the IP address of the Unity Express is required in order for inbound traffic to come in from
the network.
If you are going to use web interface then you must define the HTTP Server.
For Voice Mail pilot number you must define a SIP based Dial Peer with CODEC G.711 u-law and DTMF
SIP NOTIFY.
Unity Express Setup (cont’d)
telephony-service
dialplan-pattern 1 44028016... extension-length 4
Voicemail 6600
secondary-dialtone 9
web admin system name cisco password cisco
dn-webedit
Call-forward pattern 66…
!
ephone-dn 1
number 5001
description 44028016001
VoiceBootcamp
call-forward busy 6600
call-forward noan 6600 timeout 10
Ephone 2
mac-address x.x.x.x
username ukphone1 password cisco
!
Ephone-dn 15 Ephone-dn 16
number 8001…. number 8000….
mwi on mwi off
215
Because CallManager Express and Unity Expres shares the same Web interface, you must allow CME
admin access to CUE module. This is done by defining a web admin account under Telephone Services.
When you log in to Unity Express for the first time you must define the username and password of the
CME along with the IP address so that CUE can be authenticated by the CME router.
Each EPHONE must have a username and password define in order for Unity Express to recognized
them as a potential users of the voicemail system. Otherwise you will have to manually create a mailbox
for each and every user
MWI numbers must be define as per the s. Otherwise CUE will not recognized them. The 4 dots you see
after the number will be used to substitute the extension number of the user who receives a new
voicemail
Unity Express Wizard
VoiceBootcamp
216
• Go to http://135.XX.67.230/ IP address of Unity Express

• Enter the username and password for unity express (not the CCME username)
Unity Express Wizard Step 2
VoiceBootcamp 217
Here you will define the IP address of the CME router and the username and password.
VoiceBootcamp 218
Under Call Handling you must define the Voicemail Pilot Number and MWI Numbers
VoiceBootcamp 219
Select the extension and users who’s mailbox you want to create.
NOTE: Some time if you check the Set CFNA/CFB process may hang. It is suggested that you manually
set the call forward busy or no answer per ephone-dn before coming to this page. And Make sure you do
not select Set CFNA/CFB.
Select the user ID that you wish to create a mailbox for. You can make these users an administrator of
Unity Express as well.
VoiceBootcamp 220
Ever voice mail system out there in the market has a default password that is used by all the new mailbox
that are created. This way when a employee join a company and he/she gets her extension, they login to
voicemail using the default password and then system prompt them to change it.
Here you decide how Unity Express will handle it. Now you can set this value to be automatically
generated or leave it blank. If you select Generate Random Password, then at the end system will show
you the entire generated User password and PIN numbers.
 Password is used by user to login to web site

 PIN is used by the user to login to their voicemail to check voice mail from the phone
You can also set some threshold value to certain parameters such as how big the mailbox size can be
etc.
Unity Express Wizard 6
VoiceBootcamp
221
Unity Express receive calls from the end users. Now when Unity Express will receive calls when
someone dial the Voicemail pilot #. For example if the voicemail pilot number 6600 and user dial 6601,
there is chance that unity express may not answer that call. Unity Express Call handling tells the unity to
play Welcome Greeting or Closed Greeting by matching the inbound DNIS number to Voicemail Number.
If inbound DNIS is not the voicemail number then unity will match it against Auto Attendant Access
Number. Voice Mail Number should be the same as Voicemail Pilot # configured in Callmanager express
VoiceBootcamp
222
Save the information and logout.

Unity Express : Phone
VoiceBootcamp
223
• From the configure menu, select phone.

• This page list all the phones that are found in CallManager Express.
• Click on the Mac address to edit/update the phone
Unity Express Phone Configuration
Block Caller ID
Directory Number
VoiceBootcamp
224
• Use this dialog box to configure the phone parameters.

Unity Express User Configuration
VoiceBootcamp 225
• Define user information.

• Primary E.164 number is often the full E.164 number such as for mailbox 6001 it should be
44028916001
• Be default Unity express use primary extension of the phone. However if phone has multiple DN,
then select the primary extension from the list.
• You can generate password or manually configure one.
Unity Express Mailbox
VoiceBootcamp
226
• Define mailbox size, caller message size in seconds etc.

Cisco Unified Contact
Center Express
VoiceBootcamp 227
UCCX
• Cisco CallManager (CCM)

–Implementation of IP Phones, directs VoIP calls to UCCX Express
• Directory (LDAP)
–Stores UCCX Express configuration data and UCCX Express scripts
• UCCX Express Server
–Runs Engine
• UCCX Express Script Editor
–Create and update UCCX Express scripts
• Cisco Agent Desktop (CAD)/
Cisco Supervisor Desktop (CSD)
VoiceBootcamp
–Agent and Supervisor functions
Agent Monitoring and recording
228
• Maybe add better descriptions here and split across twos ???
UCCX Call Flow
JTAPI Provider = CCM IP address
RmCM Provider = CCM IP Address Normal user
Trigger is registered with CCM as a CTI Route Point
CTI Port JTAPI CTI Route

Known as Call user Point
Control Group 3500
UCCX DN – 3201 - 3203 CCM
3500
Trigger
RmCM
Cisco Media User
Application
3001 5001
VoiceBootcamp
A1 A2
Resource
Script Queue Group/Skills
CSQ
229
UCCX Express server requires an administrative account which must be created in CallManager as
normal users. Once this user is created, user must login to UCCX Express using the default
Administrator account and run the setup then define the new username as a new administrative account.
Communication between UCCX Express and CallManager is controlled by JTAPI interface. For this you
must create a user name that will act as JTAPI users. Now Jtapi user controls the CTI Ports and CTI
Route Points which is used by UCCX Express server to send/receive calls. CTI route point will act as a
Trigger while CTI Ports are used to route signaling between UCCX Express and CCM. Jtapi user must
be associated with all the CTI Ports and route point created by UCCX Express
NOTE: when creating JTAPI port or CTI Route point treated like as if it is an IP Phone in HQ. So
whatever the HQ Phone has in terms of device pool, CSS, AAR group, take those item in to
consideration. Does CTI Port require AAR Group for example? Or External Phone number mask. Some
time you may not be explicitly asked about it but you must do it anyway or may be some indirect task may
fail.
Once JTAP is integrated, configure Resource Manager User which will control the agent IP Phone. Now
RM user is a CCM user which must be manually associated with agent IP Phone with no primary
extension or no icd extension selected. RM user is responsible for monitoring and controlling the agent
IP Phone and their status.
For each agent, a user account is created as well. The difference between agent account and RM user is
agent account will be associated with respective agent Phone with primary extension and ICD Extension
selected.
For VOICE or RTP path, UCCX Express must have Cisco Primary Dialogue (Cisco Media) which defines
how many RTP session can be establish from UCCX server. Usually for lab purpose, we define Cisco
Media port to be equal as Call Control Port which is used for signaling.
You must also create your Resource group or skills and assign them to the Agents. Each agent must
belong to a resources group or skills before they can start receiving a call from the queue.
Contact Service Queue (CSQ) must be created in order to define the ICD Script to route calls to an agent.
If CSQ is not defined then call will fail if normal ICD Script is used.
Define an application which will tied to a script (such as ICD.aef) then script is tied to CSQ which in return
is tied to Resource Group. Now whoever is logged in to the resource group as an agent will be able to
receive calls from the queue. Associate a Trigger with this application.
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UCCX Express Setup
• Define a username in CallManager call “crsadmin” with

password “cisco”
• UCCX Express does not have any administrative account
therefore use the setup account to run the initial setup
– Setup User id: Administrator (A is capital)
– Setup Password: ciscocisco
• After running the wizard, select the new administrative

account and re-login to UCCX with new account
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UCCX Express will communicate with CallManager therefore all authentication is controlled from
CallManager server. First step is to configure UCCX Express and define a new administrative account.
Create a username and password in CallManager. Then login to UCCX Express server and use the
default username and password
User id: Administrator (A is capital)

Password: ciscocisco
After login in to the UCCX Express server, define the license file and LDAP information. Then select the
new users.
Step 1 UCCX Express Setup
• Step 1 – Create a Admin account in CallManager
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• When creating an account always ensure that Enable CTI Application Use is selected.
• This must be an end user
Step 2 Login to UCCX Server
Username: Administrator
Password: ciscocisco
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• When creating an account always ensure that Enable CTI Application Use is selected.
Step 3 – Define AXL User/Server
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AXL Server is the IP address of Unified Communication Manager. AXL user can be an Application user
with necessary permission.
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Step 4 – Continue After license File to activate

the component
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• Click Next
• NOTE: THERE IS A CHANCE THAT AT THIS POINT SYSTEM MIGHT CRASH. IF THAT IS THE
CASE, REBOOT THE PC
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Unified CM configuration
235
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Language Selection & User Selection
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• Select the Language and click Next

• User Configuration you will select the New Administrator. If you don’t see anything here then either
your LDAP configuration is incorrect and/or username was not created in CallManager or in LDAP
UCCX Administration Page
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• Login using the new account name

Unified CM Telephony
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• From the Subsystem select Unified CM Telephony to define number of signaling port to be created (CTI
PORTS)
Cisco Media Termination DG
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Used to carry voice traffic
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• Media termination dialog group is used to carry voice traffic. If this is not configured, call may be
connected but you will not hear any voice or RTP stream.
Call Control Group
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• Call control group is like a Signaling path. Number of simultaneous communication will depend on how
much ports are available in call control group
• When call control group is created, system will create CTI Ports in Call Manager and registered it with
UCCX
Agent Account
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• Now create user account that will be used as a agent ID to login.

• Make sure User has their phone associated with it
• Also user must have a primary extension and IPCC Extension defined.
• Once user has a IPCC Extension defined, user will be listed in RmCM Resource Pages
Resource Group
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Agent can be selected based on Skills
Or Resource group
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• Resource group are used to route calls to group of Agents.

• You can create group to manage technical team. For example: Sales Group has all sales agent while
support group has all support agent.
CSQ
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• CSQ is the queue name where call will be held temporary.

• You must define at least one CSQ in order for call to be queued properlhy
• CSQ is case sensitive.
Add New IPCC Application
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• In order for client to use IPCC, you must create application. Now Application must use some sort
scripts. Agent service is not always required. It all depends on the what the script is written for.
• IPCC Express use the Application to call the script
• Parameters in the application will depend on how the script is written. Some script may require
application to pass variables while others don’t.
Add a Trigger to call the application
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• Trigger will be used to call the application created in the previous s

• Application can have more then one trigger. One Trigger can only be associated with one Application
• Trigger will be created as a CTI Route Point in CallManager and will be associated with the JTAPI user
account
• When call arrive on the trigger, call will hit the IPCC Express. IPCC looks at the trigger and realize that
it is associated with an application which in return call the script and script does what it is designed to
do.
Trigger and CTI Ports in UCM 7
Device Menu  CTI Route Points
Device Menu  Phones
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• Both CTI Route Point (Trigger) and CTI Ports (Call Control group) must be registered in Callmanager.
• If they are not required, Re-start the CRS Engine or reboot the IPCC Server. Often you may have to
restart the Callmanager in lab environment.
IP Phone Service for Agent
• Create an IP Phone Service with the following URL and

Subscribe to IP Phone
• http://X.X.X.X:6293/ipphone/jsp/sciphonexml/IPAgent
Initial.jsp
• Where X.X.X.X is the IP address of IPCC Express Server
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• IP Phone service is required for agent to login from the IP Phone. It is case sensitive.
• Each and every IP Phone must be associated with the this services.
Agent IP Phone and RM User
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• RM user must be associated with the Agent IP Phone.

User Account Association
• Crsadmin = normal IPCC Admin account

• JTAPI USER – this user is associated with CTI Ports and
CTI Route Point created by IPCC
–No Primary extension required
–No IPCC Extension Required
• RMUser – This user is associated with Agent IP Phone

–No Primary extension required
–No IPCC Extension Required
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• Agent account just as jsmith – this is the agent
– Primary extension required
– IPCC Extension Required
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• Crsadmin = normal IPCC Admin account

• JTAPI USER – this user is associated with CTI Ports and CTI Route Point created by IPCC
• No Primary extension required
• No IPCC Extension Required
• RMUser – This user is associated with Agent IP Phone
• No Primary extension required
• No IPCC Extension Required
• Agent account just as jsmith – this is the agent
• Primary extension required
• IPCC Extension Required
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Presence 7.0 Overview
• collects information about a user's availability and

communications capabilities
• facilitate presence-enabled communications for Cisco
Unified Communications and critical business
applications
• takes advantage of Session Initiation Protocol (SIP)
technology
• Cisco Unified Presence is tightly integrated with
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various desktop clients and applications
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Presence 7.0 Overview (cont’d)
• Cisco Unified Presence Modes of Operation

–Cisco Unified Communications Mode (30,000 users)
–Microsoft Office Interoperability Mode (10,000 users)
• Microsoft Outlook Calendar Integration

• Cisco Unified Presence Federation
• Centralized Communication Utility
• Cisco Personal Communicator Client (CUPC)
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Overview of Presence 7.0
Step 1
CallManager: Service Parameters set the
Set the Default Inter-Presence Group Subscription to

“Allow Subscription”
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Cisco Unified Presence is a standards-based platform that collects information about a user's availability
and communications capabilities to provide unified user presence status and facilitate presence-enabled
communications for Cisco Unified Communications and critical business applications. With this scalable
and easy-to-manage solution, Cisco Unified Presence delivers a consistent presence-enabled
communications experience across Cisco Unified Communications applications everywhere, every time,
independent of user device, application, or workspace location. In addition, Cisco Unified Presence gives
customers and partners the flexibility to presence-enable and streamline business communications by
interoperating with critical business applications through open interfaces.
Step 2 – SIP Trunk Profile
• Go to System / Security-Profile
• Add a SIP trunk security profile
• Use these settings
• Enable:
Accept Presence Subscription

Accept Out-of_Dialog REFER
Accept Unsolicited Notification
Accept Replaces Header
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• SIP Trunk security profile must have the following Item checked
• Accept Presence Subscription

• Accept Out-of_Dialog REFER
• Accept Unsolicited Notification
• Accept Replaces Header
• These settings allow Presence information to carried over the trunk line
Step 3 – CallManager:
Add SIP trunk
CallManager: Add SIP trunk

• Add a SIP trunk for each CUPS associated with this CallManager
• Device / Trunk / Add Trunk / Protocol:SIP
• Subscribe CSS – Select a CSS that has access to all Phone DNs
CUPS Server
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• Add a SIP Trunk that will be used by Presence Provider

Step 4 – CallManager: AXL user
CUPS uses AXL SOAP to access the CM database

You have to configure a username/password:
• Easy&Fast: use the CCMAdministrator user

This user has the ‘Standard AXL API access’ role
• Better: Create an application user with this role

– Add an Application user (for example “AXLuserCUPS”)
Create a group: “group_AXLaccess”

– Add this user to the “group_AXLaccess” group
– Click on the listbox (upper right corner) “Assign Role to
Usergroup”
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– Assign the “Standard AXL API access” role to this group
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• AXL user is the user account that will be used by Presence to administer and manage settings in
CallManager.
• Create a new user account and ensure it has appropriate level of permission.
Step 5 – CallManager: Services
Check if the following services are activated and

running. These are required for CUPS to operate
Cisco CallManager
Cisco TFTP
Cisco Extension Mobility
Cisco CallManager Cisco IP Phone Services
Cisco AXL Web Service
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• In order for Presence server to work properly, make sure the above services are running properly.
Step 6 – Call Manager
Configuration for IPPM
• Add an IP Phone if it doesn’t exists for a user.

• Add users to the following group ‘Standard CCM Enduser’ and
‘Standard CTI Enabled’ group.
• Associate the End User to their Device (IP PPhone)
• Have to set the Primary Extension for persistent login
where you only have to enter the PIN
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• Cisco IP Phone Messenger enables your Cisco Unified IP phone to receive, send, and reply to instant
messages
• It is only available if Presence is deployed.
• The Cisco IP Phone Messenger service is an application that runs on your Cisco Unified IP Phone
• A service is a special type of XML-based application that can run on Cisco Unified IP Phones
• Service might be assigned to a phone associated with your user ID (assigned) or not associated
(unassigned)
Add PhoneMessenger user
CallManager: Add PhoneMessenger user

• Add an application user called “PhoneMessenger”
• Associate all phones that are going to use IPPM
• Put this user in the ‘Standard CCM End User’ group
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Phone Messenger is another type of Presence client that allows users to send text message over IP
Phone using IP Phone Service. In order for Presence to use this feature with Callmanager, it must
authentication itself to Callmanager. Therefore you must create an Application User in UCM with
Standard CCM End user Groups.
Add XML service
Add the IP Phone Messenger (IPPM) XML service

• Service name: IPPM
Service description: IP Phone Messenger
URL: http://<servername>:8081/ippm/default?name=#DEVICENAME#
• Use the IP address instead of <servername> if DNS is not enabled on the
phone
• No parameters are needed
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 This is the IP Phone service that needs to be subscribed by all the end point that will use the IP
Phone messenger service.
 Use the Presence Server IP address in the URL field.
Step 9 – CallManager: Subscribe phones
• Phones that are going to use IPPM now have to subscribe to the IPPM XML
service and Reset the phones
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 Subscribe the IP phone to this URL

 URL: http://<servername>:8081/ippm/default?name=#DEVICENAME#
Step 10 – Call Manager:
Capabilities Assignment
• Go to the CallManager admin GUI
• Set the capabilities for each user
• Use Bulk Assignment for >1 user
• Licenses are used based on capabilities
• Select Enable UPS
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• When Presence is deployed, you must decide which user will have presence capability.
Not all user
required presence enable. So depending on the requirement from company’s policy, Presence feature
must be able enable per user basis.
Step 11 – Call Manager Configuration
for Cisco Unified Personal
Communicator
Add an IP phone (if none exists) for the user

• Add the primary extension as one of the IP phone’s lines -
this is the only line CUPC can control
• Create the end user account, and designate the user’s
primary extension (DN), if not already configured
• Associate the user with the IP phone, and enable CTI
control for the phone
• Add the user to the “Standard CTI Enabled” and “Standard
CCM End Users” groups
• Configure Digest Credentials for the user
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• CUPC client is the client that is used by End user to login to CallManager and see presence status of
the other users. Think of CUPC client is like Microsoft Messenger but only works with Cisco.
• Using a single application you can make a voice call, video call, web conference, check your voice mail,
chat with someone etc.
• CUPC client require license for every users. It must be associated with a Hardware IP phone and/or
softphone by using user and owner relationships.
Step 12 – Call Manager Configuration for
CUPC – Add CUPC Device (cont’dt)
• Create a “Cisco Unified Personal Communicator” softphone device - name MUST

be: UPC<uppercase-userid>
•More on the naming scheme on separate slide

•Configure a single DN (use the primary extension shared with the IP phone)
•Disable “Allow control of device from CTI” under CUPC Device
•Associate the user with the CUPC device
•Configure voicemail settings for the shared line, if not already configured
•SIP Phone Security Profile – Select Standard SIP Profile for Auto-Registration
•SIP Phone – Select Standard SIP Profile
•Digest User – Select the user id who will be associated with this device
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• When adding CUPC Client Device in Device Menu, Ensure that Device Name starts with UPC follow by
the username. For example if your username is: cisco then device name for CUPC Client is:
UPCCISCO (all in capital letters)
Step 13 Call Manager Configuration for
CUPC – Add CUPC Device
• Create a softphone device –

use “Cisco Unified Personal
Communicator”
• The Device Name MUST be the
user’s ID in the form;
UPC[0-9A-Z]{1,12}
meaning that you take out
anything that not’s a letter or
number and only use the first
12 characters and capitalize
them
• Example: user id “fkhan”
becomes; UPCFKHAN
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User id – VOICEBOOTCAMP
becomes UPCVOICEBOOTCAMP
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 Add New Device (Device  Phone  Add New Phone)

 Device Type will be Cisco Unified Personal Communicator
 Device name is UPC follow by username all in capital letter.
Step 14 CUPS CONFIGURATION
Define username
And password of
AXL User
Ie. Administrator
Define the hostname of UCM
And IP address of
CallManger 7.0
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• Cisco Unified Presence is dependent upon Cisco Unified Communications Manager for configuration of
users, devices, and licensing. The Cisco Unified Presence publisher communicates with the Cisco
Unified Communications Manager publisher via the AVVID XML Layer Application Programming
Interface (AXL API)
• If AXL username and password is not correct Sync Agent will not start
• User must have access to Standard AXL API
Step 15 CUPS CONFIGURATION
Define username
And password of
AXL User
Ie. Administrator
Security Key defined
During installation
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• Enter the security key that was provided during the installation.
Step 16 Licencing
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• Check to see if you have enough license for CUP server.

• You need at least one Unity for One server
Step 17 Service activation
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• All services must be running

Step 18 – Add Presence Gateway
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• Presence gateway is required to push and pull all request of the users status.
• Presence gateway in this case is going to CallManager
Step 19 Application Listener
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• You can configure application listeners for the SIP proxy server, presence engine, and profile agent.
The system binds each application listener to a specific address and port combination. If you choose
TLS protocol, you must also choose a TLS context
• You must restart the SIP proxy server before any changes that you make to the application listeners
take effect. To restart the proxy server, select Presence > Routing > Settings
• For Cisco Proxy Server listeners, there is a limit of 20 listeners.
Step 19 Application Listener
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• In the Incoming and Outgoing Access Control List (ACL), you can configure patterns that control which
incoming hosts and domains can access Cisco Unified Presence without authentication. Cisco Unified
Presence accepts a range of IP address patterns in addition to fully qualified names of incoming hosts
or domains. The Allow directive followed by "from" determines which hosts can access the server.
• All hosts - Allow from all = all

• A partial domain name = voicebootcamp.com
• Based on IP address = 192.168.1.0/24
• Configure an address which will be added to the SIP Proxy list of allowed incoming and outgoing
addresses
• Any address added to this list will bypass digest authentication
• By default, system behavior is to deny all incoming and outgoing requests
Step 20 service parameters
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 Ensure all the parameters are configured properly such as default PROXY Domain name etc.
Step 21 SIP Proxy Settings
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• SIP Proxy Domain must be defined.

How users will login such as: users@135.2.100.22 or
users@voicebootcamp.com or users@presence.voicebootcamp.com
Step 22 IP PhoneMessenger
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• The Cisco IP Phone Messenger service, included with Cisco Unified Presence, provides an Instant
Messaging (IM) client on Cisco Unified IP Phones with availability-enabled contacts lists. This feature
integration with Cisco Unified Presence gives phone users who might be away from their computers a
quick way to check on the availability status of colleagues. As well as real-time collaboration
capabilities, the feature allows users to send and receive short text messages, many of which are
preinstalled in a list of commonly used phrases and full sentences that users can select rather than
enter on the phone keypad. Message recipients can reply to their messages or press the Dial softkey to
call back without having to look up or dial the number
Step 23 Presence Settings
Select the UCM SIP Trunk

Created in CallManager
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• In Presence setting you must define which SIP Trunk to use from CallManager for Presence.
Step 24 CUPC Settings
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• Here you define some parameters for CUPC Client such as when CUPC Clients logins which TFTP
server they will get all the necessary files.
• If you are defining Active Directory you can define certain parameters like what should be used as a
User ID from Presence client. By default it users SAM Account
Step 24 CUPC Settings
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• If you need to connect to Microsoft OCS for example then you define the CTI Gateway.
Step 25 Adding Voicemail
Unity Connection Server Configuration

• Go to Application – Unified Personal Communicator – Unity Server – Add New
• Name – Hostname of Unity Connection Server
• IP Address – IP Address of Unity Connection Server
• Port – 143
• Protocol Type - UDP
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It is IMAP Port 143
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• In order for presence to access Voicemail, ensure the Unity Connection IP address along with port
number
CUPS Configuration for CUPC – Unity Server
Profile
Unity Server Profile Configuration

• Go to Application – Unified Personal Communicator – Unity Profile – Add
New
• Name – UnityConnection (Name can be any name)
• Voice Messaging Pilot – UnityCon
• Primary Unity Server – Select Unity Connection Server that was added
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• Make sure all the users that require this voicemail profile must be included in the settings
CUPS Configuration for CUPC – Meeting
Place Express Server
Meeting Place Express Server Configuration

• Go to Application – Unified Personal Communicator – Meeting Place Server –
Add New
• Name – Hostname of MPX Server
• IP Address – IP Address of MPX Server
• Port – 80
• Protocol Type – HTTP/HTTPS
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• Ensure that MeetingPlace is working properly.

CUPS Configuration for CUPC – Meeting Place
Profile Configuration
Meeting Place Profile Configuration

• Go to Application – Unified Personal Communicator – Meeting Place Profile –
Add New
• Name – Profile Name
• Primary MeetingPlace Server – Select Meeting Place Server that was added
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 Enter the Meeting Profile information and associate the users to this profile. Only these users will
have access to MeetingPlace Express.
CUPS Configuration for CUPC – CTI Gateway
CTI Gateway Server Configuration

• Go to Application – Unified Personal Communicator – CTI Gateway Server – Add
New
• Name – Hostname of Server running CTI Service (One of the CCM Server)
• IP Address – IP Address of Server running CTI Service Server
• Port – 2748
• Protocol Type – TCP
Add additional CTI Gateway Server if more than one CTI Server is available
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 If CTI Gateway is required then enter the CTI Gateway information here.
CUPS Configuration for CUPC – CTI Gateway
Profile
CTI Gateway Server Profile Configuration

• Go to Application – Unified Personal Communicator – CTI Gateway Server
Profile – Add New
• Primary CTI Gateway Server – Select Primary CTI Gateway server
• Backup CTI Gateway Server – Select Backup CTI Gateway Server if any
configured
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 Create a CTI Gateway profile and associate this profile to users.

CUPS Configuration for CUPC – LDAP Server
LDAP Server Configuration

• Go to Application – Unified Personal
Communicator – LDAP Server – Add New
• Name – Hostname of LDAP Server
• IP Address – IP Address of LDAP Server
• Port – 389
* LDAP Server should be V3 compatible &
anonymous read access is sufficient
LDAP Server Configuration if

Global Catalogue server is used
for LDAP.
• Port – 3268
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Please use the above Port and
Protocol type if Global Catalogue
server is used for LDAP.
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 If CUPC Users require access to corporate directory to search for contacts then LDAP must be
used.
 Create a LDAP host configuration based on your existing AD schema
CUPS Configuration for CUPC – LDAP Profile
LDAP Profile Configuration

• Go to Application – Unified Personal
Communicator – LDAP Profile – Add
New
• Distinguished Name, Configuration
Name and PWD – Fill any Value
• Distinguished Name – Enter user-id
with read access to LDAP.
• Password – Enter Password.
• Un-Check Anonymous Bind
• Set search Context – Set O and OU,
OU should contain users. Example
show in the picture is for AD.
• Primary LDAP Server – Select LDAP
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Server that was added.
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 Create a LDAP profile and associate it with all the users.

QOS
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QoS
•L2/L3 classifications and policing

•Queuing mechanisms
•LFI
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 QoS will focus on the voice related configurations. Although QoS is a full topic and require a
separate class to complete and cover all the topics, in this session we will focus on the voice
portion.
 For QoS please read the QoS SRND Guide from www.cisco.com/go/srnd
Enabling QoS in the Campus
Pream. SFD DA SA Type 4 TAG

Bytes PT Data FCS
Ethernet Frame
hree Bits Used for CoS
802.1p User Priority) 802.1Q/p
PRI CFI VLAN ID
Header
CoS Application
• 802.1p user priority field also 7 Reserved
called Class of Service (CoS) 6 Reserved
• Different types of traffic are 5 Voice Bearer
assigned different CoS values 4 Video Conferencing*
• CoS six and seven are reserved 3 Call Signaling
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for network use 2 High Priority Data
1 Medium Priority
Data
* Including Audio and Video 0 Best Effort Data
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• Voice traffic can be identified in many ways. The easiest way to identify voice traffic is to have the end
device (the IP phone or gateway) mark its traffic appropriately.
• Cisco IP phones tag their bearer traffic at Layer 2 with a CoS of 5 and set the Layer 3 DSCP marking to
EF.
Campus QoS Considerations
Switch Detect IP Phone
1 Switch trust IP phone PC VLAN = 10
Phone VLAN = 110
TRUST BOUNDARY
4
2
“CoS 5 = DSCP 46” “Voice = 5, Signaling = 3”
“CoS 3 = DSCP 24”
“CoS 0 = DSCP 0” 3
All PC Traffic Is Reset to CoS 0 PC Sets CoS to 5 for All Traffic
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 Typical Cisco IP Phone will generate traffic with the following tag
o Signaling – CS3 or DSCP 24
o Media/RTP – EF or DSCP 46
o PC Traffic will be over written by the IP Phone to 0
Classification and Marking Design
QoS Baseline Marking Recommendations
L3 Classification L2
Application
IPP PHB DSCP CoS
Routing 6 CS6 48 6
Voice 5 EF 46 5
Video Conferencing 4 AF41 34 4
Streaming Video 4 CS4 32 4
Mission-Critical Data 3 AF31* 26 3
Call Signaling 3 CS3* 24 3
Transactional Data 2 AF21 18 2
Network Management 2 CS2 16 2

Bulk Data 1 AF11 10 1
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Scavenger 1 CS1 8 1
Best Effort 0 0 0 0
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• The richness of Cisco's QoS feature set presents a myriad of deployment options and combinations,
which nearly every QoS-savvy engineer has a slightly different opinion on how best to enable.
• Therefore, to present a consistent QoS story, Cisco has adopted a new initiative called the “QoS
Baseline.” The QoS Baseline is a strategic document designed to unify QoS within Cisco, from
enterprise to service provider and from engineering to marketing. The QoS Baseline was written by
Cisco's most qualified QoS experts.
• The QoS Baseline specifies 11 traffic classes within the enterprise. An important note is that the QoS
Baseline is not dictating that every enterprise deploy 11 different traffic classes immediately (see
following for more details), but rather it is considering enterprise QoS needs of not only today, but also
the foreseeable future. Even if an enterprise needs to provision for only a handful of these 11 classes
today, following QoS Baseline recommendations will enable them to leave options open for smoothly
provisioning additional traffic classes in the future.
• Note: The QoS Baseline recommends marking Call-Signaling to CS3. Currently, however, all Cisco IP
Telephony products mark Call-Signaling to AF31. A marking migration from AF31 to CS3 is planned
within Cisco, but in the interim it is recommended that both AF31 and CS3 be reserved for Call-
Signaling and that Locally-Defined Mission-Critical data applications be marked to DSCP 25. Upon
completion of the migration, the QoS Baseline marking recommendations of CS3 for Call-Signaling and
AF31 for Locally-Defined Mission-Critical applications should be used. These marking
recommendations are more inline with RFC 2597 and RFC 2474.
Cisco Catalyst 3550 QoS Design
Enable switch-wide QoS.
Cat2(config)#mls qos
Modify the default CoS-to-ToS mapping table. You must setup a translation between CoS and DSCP because there
at
only 8 CoS labels and 64 possible DSCP labels. The default mapping table looks like
Cat2#show mls qos maps

Cos-dscp map:
cos: 0 1 2 3 4 5 6 7
--------------------------------
dscp: 0 8 16 24 32 40 48 56
Change the defaults so that:

• CoS 3 maps to CS3 (24)
• CoS 4 maps to AF41 (34)
• CoS 5 to EF (46)
Cat2(config)#mls qos map cos-dscp 0 8 16 24 34 46 48 56
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Cat2#show mls qos maps
Cos-dscp map:
cos: 0 1 2 3 4 5 6 7
--------------------------------
dscp: 0 8 16 24 34 46 48 56
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Often some catalyst switch QoS may not be enabled by default. You must enable the qos on the switch.
Default mapping may not reflect the correct settings of CoS to DSCP mapping. Therefore mls qos map
command should be used.
For all Signanling traffic that exceed 39K
mark down the DSCP to 8 – Cat3550
CAT2(config)#mls qos map policed-dscp 0 24 46 to 8  ! Excess traffic marked 0 or CS3 or EF will be remarked to CS1
CAT2(config)#
CAT2(config-cmap)#class-map match-all SIGNALING
CAT2(config-cmap)# match access-group name ACL_SIGNALING
CAT2(config-cmap)#exit
CAT2(config)#
CAT2(config)#policy-map VOICE-CONTROL
CAT2(config-pmap-c)#class SIGNALING
CAT2(config-pmap-c)# set ip dscp 24  ! Signaling is marked to DSCP CS3
CAT2(config-pmap-c)# police 39000 8000 exceed-action policed-dscp-transmit
CAT2(config-pmap-c)#class class-default
CAT2(config-pmap-c)# set ip dscp 0
CAT2(config)#
CAT2(config)#interface FastEthernet0/1
CAT2(config-if)# service-policy input VOICE-CONTROL
CAT2(config-if)#exit
CAT2(config)#
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CAT2(config-ext-nacl)#ip access list extended ACL_SIGNALING
CAT2(config-ext-nacl)# permit tcp any any range 5000 5002
CAT2(config-ext-nacl)#end
CAT2#
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In order to ensure proper traffic flow, policing may be required. In this example we are advising the
switch that if signaling traffic exceed it configured value then re-mark the traffic with a lower CS value
however switch is not dropping packet in this case. It is simply remarking the packet and sending the
packet to the next hop.
Now next hop may be a router. If so then router can be configured to drop traffic of lower CS value first
should there be any congestion in the network.
VoiceBootcamp
Classify Traffic using Access-List
Cat2(config)#ip access-list extended VOICE

Cat2(config-ext-nacl)#remark Match the UDP ports that VoIP Uses for Bearer Traffic
Cat2(config-ext-nacl)#permit udp any any range 16384 32767
Cat2(config)#ip access-list extended VOICE-CONTROL

Cat2(config-ext-nacl)#remark Match VoIP Control Traffic
Cat2(config-ext-nacl)#remark SCCP
Cat2(config-ext-nacl)#permit tcp any any range 5000 5002
Cat2(config-ext-nacl)#remark H323 Fast Start
Cat2(config-ext-nacl)#permit tcp any any eq 1720
Cat2(config-ext-nacl)#remark H323 Slow Start - Verify could be in 3000 range for CM or 11000 to 65535 with newer IOS's
Cat2(config-ext-nacl)#permit tcp any any range 11000 11999
Cat2(config-ext-nacl)#remark H323 MGCP
Cat2(config-ext-nacl)#permit udp any any eq 2427
Cat2(config-ext-nacl)#permit tcp any any eq 2428
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297
 List of Port to remember in order to create access list based on certain traffic type.
WAN Edge QoS Design
Considerations
Campus
Distribution/Core
Queuing/Dropping/
Switches Shaping/Link-Efficiency
Policies
for Campus-to-Branch Traffic
WAN Aggregator
WAN
LAN Edges WAN Edges
VoiceBootcamp 298
• A fundamental principle of economics states that the more scarce a resource, the more efficiently it
should be managed. In an enterprise network infrastructure, bandwidth is the prime resource and it is
scarcest over the WAN. Therefore, the case for efficient bandwidth optimization via QoS technologies is
strongest over the WAN, especially for enterprises that are converging their voice, video, and data
networks.
• This chapter provides design guidance for enabling QoS over the WAN. It is important to note that the
recommendations put forward in this chapter are not autonomous. They are critically dependant on the
recommendations discussed in Chapter 2, “QoS in an AVVID-Enabled Campus Network.”
• This chapter focuses strictly on the WAN components of the Cisco AVVID Network Infrastructure,
specifically the:
• WAN aggregation routers
• Remote-branch routers
• WAN media
WAN Edge QoS Design
Considerations
• Slow-speed links (≤ 768 kbps)

–Voice or video (not both)—3 to 5 class model
–LFI mechanism required
–cRTP recommended
• Medium-speed links (≤ T1/E1)
–Voice or video (not both)—5 Class model
–cRTP optional
• High-speed links (> T1/E1)
–Voice and/or video—5 to 11 Class (QoS baseline) model
VoiceBootcamp
–Multiple links require bundling or load-balancing
–Very high-speed links (DS-3/OC-3) require newer CPUs
299
 CRTP, or RTP header compression, is a method for decreasing the size of the Voice over IP (VoIP)
packet headers to reduce the bandwidth consumed
 CRTP was designed for reliable point‐to‐point links with short delays
WAN Edge Bandwidth Allocation Models
Three-Class (VoIP and Data Only) WAN Edge Model
Best
Effort Voice
(62%) 33%
VoiceBootcamp
Call-
Signaling
5%
300
Example of 3 class model QoS.

WAN Edge Bandwidth Allocation Models
Three-Class WAN Edge Model Configuration Example
!
class-map match-all VOICE
match ip dscp ef ! IP Phones mark Voice to EF
class-map match-any CALL-SIGNALING
match ip dscp cs3 ! Call-Signaling marking (new)
match ip dscp af31 ! Call-Signaling marking (old)
!
!
policy-map WAN-EDGE
class VOICE
priority percent 33 ! Recommended to keep LLQ ≤ 33%
compress header ip rtp ! Optional: Enables Class-Based cRTP
class CALL-SIGNALING
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bandwidth percent 5 ! Minimal BW guarantee for Call-Signaling
class class-default
fair-queue ! All other data gets fair-queuing
!
301
Class Map is used to classified the inbound traffic to a specific class. This class can then be reference in
the Policy map where re-classification or modification is done.
Frame Relay QoS Design
FRTS (+ FRF.12) Recommended
Parameters Table
PVC Fragment
CIR Bc
Speed Size
56 kbps 53500 bps 532 bits per Tc 70 Bytes
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302
 Standard table for Fragment calculation based on given CIR.

Frame Relay QoS Design
WAG FR Link ≤ 768 kbps

<a 3 to 5 Class Model can be used>
Optional: Enabling Class-Based cRTP Frame
Relay
Cloud
! BR
policy-map MyQoS
class class-default
shape average 729600 7296 0 ! CIR=95% rate, Bc=CIR/100, Be=0
service-policy WAN-EDGE ! Queues packets before shaping
!
!
interface Serial2/0
no ip address
encapsulation frame-relay
!
VoiceBootcamp
interface Serial2/0.12 point-to-point
frame-relay interface-dlci 102
class MyQoS-VOIP ! Binds the map-class to the FR DLCI
!
!
map-class frame-relay MyQoS-VOIP
service-policy output MyQoS ! Attaches MQC policies to FR map-class
frame-relay fragment 480 ! Enables FRF.12
!
303
 In order to enable FRF 12 ensure frame relay class map has frame-relay fragment command with
the right value
MLPoFR QoS Design
<a 3 to 5 Class Model can be used> WAG MLPoFR Link ≤ 768

Optional: Enabling Class-Based cRTP kbps
! Frame
interface Serial6/0 Relay
no ip address Cloud
encapsulation frame-relay BR
frame-relay traffic-shaping
!
interface Serial6/0.60 point-to-point
bandwidth 256
frame-relay interface-dlci 60 ppp Virtual-Template60 ! Enables MLPoFR
class FRTS-256kbps ! Binds the map-class to the FR DLCI
!
interface Virtual-Template60
bandwidth 256
ip address 10.500.60.2 255.255.255.252
service-policy output WAN-EDGE ! Attaches MQC policy to map-class
ppp multilink
VoiceBootcamp
ppp multilink fragment-delay 10 ! Enables MLP fragmentation
ppp multilink interleave ! Enables MLP interleaving
!
map-class frame-relay FRTS-256kbps
frame-relay cir 243500 ! CIR is set to 95% of FR DLCI rate
frame-relay bc 2432 ! Bc is set to CIR/100
frame-relay be 0 ! Be is set to 0
frame-relay mincir 243500 ! MinCIR is set to CIR
no frame-relay adaptive-shaping ! Adaptive shaping is disabled
! 304
Unified Mobility
VoiceBootcamp 305
Cisco CallManager Extension Mobility Overview
•
• Log on to a Cisco IP Phone 7940 or 7960
in a Cisco CallManager cluster to get User Office
extension Single IP Phone 7960
(x5000)
• Device profile includes: extension, Cluste
services, class of service restrictions
applied to r LDAP
Directo
IP Phone IP
IP LAN ry
Phone
• Login modes: Service
– Auto-logout other s CRA
IP Phones Server
7960
– Keep login on other IP Phones
User Logged On to Phone
• Logout modes: (Device Profile with x5000)
VoiceBootcamp
– Explicit logout at IP Phone
– Timed logout
306
The Cisco CallManager Extension Mobility feature allows users to temporarily access their
Cisco IP Phone configuration such as their line appearances, services, and speed dials from other
Cisco IP Phones.
With Cisco CallManager 4.0, extension mobility functionality extends to most Cisco IP Phone models
and you can configure each Cisco IP Phone model to support Cisco CallManager Extension Mobility.
This allows users who do not have a user device profile for a particular Cisco IP Phone model to use
Cisco CallManager Extension Mobility with that phone model.
Cisco CallManager Extension Mobility Service
Parameters Configuration
VoiceBootcamp
307
• Under Service parameter you must define some EM parameters such as allow multiple login
Device Profile Default Configuration for the
Cisco IP Phone 7960
VoiceBootcamp
308
Create a extension mobility device profile. Should match the phone type. For example if the phone you
want to use as a extension mobility phone is a 7961 then device profile should be 7961 as well.
User Profile Creation
VoiceBootcamp 309
 Every user that will use extension mobility must have a device profile associated. You can have
multiple device profile per user and designate a single Device profile for default or let the user
select by them self during login (in this case do not select default)
Unified Mobility
Single Number
Reach
VoiceBootcamp
310
Cisco Unified Mobility
Cisco Unified Customer Mobile Voice

Call to
Communication Access
Mobile
s Manager establishes a
Call to Voice
Access system to create
office enterprise calls
number. directory
number. from any
PSTN location.
Gateway
Mobile Connect
Remote lets remote and
Phone office phones
Office ring
Phone simultaneously.
–Cisco Unified Mobility has two components: Mobile Connect and Mobile Voice
Access (MVA).
VoiceBootcamp
–With Mobile Connect, calls placed to office phones ring the office phones and
associated remote phone.
–MVA allows users to call into the enterprise from any phone and place outgoing
calls that appear to come from their office phone.
311
 Mobile connect allows you to ring multiple device when someone call your extensions
simultaneously
 MVA is on the other hand allows you to dial a access number to call your corporate office and
once authenticated it will allow you to dial anywhere else as if you are in dialing from office.
 Authentication is done based on one of the single number reach remote destination number
Cisco Unified Mobility Features
–Single (office) number for multiple devices:

•Enterprise caller ID preservation
•Single enterprise voice mailbox
–User-configurable access lists to permit or deny calling numbers
that can ring a specific remote phone
–User interface to enable or disable Cisco Unified Mobility:
•Mobile Voice Access TUI
•Cisco Unified Communications Manager user webpages
VoiceBootcamp
–Access to enterprise features from remote phones using DTMF:
•Softkeys can be used on phones with smart client installed.
–Call logging (CDR)
312
Mobile Connect Call Flow—Incoming Calls to
Office Phone
Cisco Unified
Communication Outside
s Manager Caller
Gateway
514- 416-
555- 555-
3XXX 1555 Call to
1-514-555-
Mobile 3001
Connect
PSTN
Caller ID:
416-555-1555
Office 604-555-
Phone 2002 Remote
3001 Phone of
VoiceBootcamp
3001
Outside caller calls office phone 3001 (dials 1-514-555-3001).
Mobile Connect rings office phone and remote phone.
Call is picked up at remote phone; caller ID of outside caller is
preserved at remote phone.
313
 Incoming call arrives on your IP Phone. UCM is monitoring all the activity on the line.
 It place an outbound call to remote destination number configured for this IP Phone
Cisco Unified Mobility Configuration
Elements
Configuration
Configuration Element
Element
Function
Name
The end user is referenced by the office phone and remote destination
End User profile. Mobile Connect and/or MVA must be enabled.
A maximum number of remote destinations can be configured.
Phone The office phone needs to be configured with an owner (i.e., the end user).
Remote A virtual phone device. Per office phone number, a shared line is
Destination configured. End user, (device) CSSs, and MOH audio sources are
Profile specified. One or more remote destinations are added.
Associated with shared line(s) of remote destination profile. Configured
VoiceBootcamp
Remote
with destination number. Optionally, access lists can be applied. Mobile
Destination
Phone and Mobile Connect functions are selectively enabled.
Filters used to permit or deny incoming calls placed to the office phone
Access List
to ring a remote destination. Permitted or denied caller IDs are specified.
MVA Media Media resource used to interact with the VoiceXML call application running
Resource on a Cisco IOS router. Only required for MVA.
314
Shared Line Between Phone and
Remote Destination Profile
Line1: 3001 Office Phone 1 Line1: 3002 Office Phone 2

Partition MAC Address Partition MAC Address
CSS Owner CSS Owner
etc. CSS etc. CSS
etc. etc.
shared line shared line
Call to
shared line Remote Line1: 3001 Remote
rings office Destination1 Partition Destination
phone line : CSS Profile
and remote 914168391717 etc. User ID
destination(s
Remote Line2: 3002 CSS
) associated
VoiceBootcamp
Destination2: Partition Rerouting CSS
with etc.
9011971380523 CSS
corresponding
0 etc.
line of
remote
destination
profile. 315
 Remote Destination Profile is like a virtual phone of the actual physical/soft IP Phone. Remote
Destination Profile or RDP can have more than one line and each line pointing to different remote
devices.
 Remote destination number (RDN) is the actual number of the device where CallManager will
send the calls to. RDN can be cell phone, home phone etc. Each RDN is associated with a RDP
and associated with a line in that RDP
Associate user account to a phone for
mobility
VoiceBootcamp
316
 Each user who requires mobility solution must have their username associated with respective
devices as well as designated as the owner of that device.
Remote Destination Profile
Extension of
The user whose
Desk phone
Will be monitored
VoiceBootcamp
One of the destination where phone will ring when
Some one is calling the user at their desk phone
317
• Remote Destination Profile is like a virtual phone which is associated with the users main desk phone
or softphone and multiple destination.
• Extension must be the same as user’s physical device or softphone that will be monitored
• Remote Destination are the numbers where the phone will ring simultaneously
Remote Destination Info
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318
• Answer Too Soon Timer means that this is the minimum number of ms must pass before a Mobile
Phone can answer
• Answer too Late Timer means that this is the minimum number of ms must pass before a mobile phone
must answer
• Delay Before Ringing Timers means that system must wait this timer before start ringing the Mobile
phone
• Mobile Phone means allow call to be transfer to mobile phone
• Enable Mobile connect means when deskphone received call, it must ring the mobile phone as well
Tips and Stretagy
VoiceBootcamp 319
Exam Tips
• Read the entire exam

• Redraw your topology
• Time management
• Clarification
• Make notes
• Check list
• Unexpected items
• Troubleshooting (10-minute rule)
VoiceBootcamp
• Functionality testing
320
Test-Taking Strategies
• Arrive early or visit the site the day before

• Don’t schedule flights too close to the
end of the exam—it can run overtime
• Get some sleep the night before the exam
VoiceBootcamp
321
Test-Taking Strategies (Cont.)
• Use question point values to judge time

• Read through the entire test first to check
for addressing issues
• Draw up a plan on which configs can be completed at the
begingin
• For example: Location, Device Pool, CSS/Partition can be
created in advanced so when you configuring phone, all
these item will be available
VoiceBootcamp
322
• Do each question as a unit—configure and verify before

moving to the next question. But do not spend more then
5 minute in troubleshooting
• Don’t assume requirements that aren’t mentioned in a
question
• Don’t make any drastic changes in the last half hour of the
exam
• Save your configs often
VoiceBootcamp 323
Troubleshooting Strategy
• Typos are the most common cause of problems found
during the lab exam
• Verify each question to ensure it is working before moving
on to other questions. This will assure you that you can
move on without any problem left behind. If everything
was working and after you have configured a new section
or question you notice a failure on your exam, you will
know exactly what is the cause of the failure.
• Keep saving your configurations before moving on to
another question. If all else fails, you can always reload a
VoiceBootcamp
device and work on something else while it comes back
up in a known state.
324
Time Management
• Total = 8 hrs = 480 mins

• Read Lab - 8 mins
• Infrastructure - 15 mins
• CM basics - 15 mins
• CME basics - 15 mins
• Gws - 15 mins
• GK - 15 mins
• Dial peers on CME and SRST - 20 mins
• SRST - 15 mins
• ---140 mins---
VoiceBootcamp
• Register Phones - 15 mins
• Media - 15 mins
325
Time Management
• ---170 mins---
• Unity/Express - 35 mins
• CRS - 30 mins
• ---220 mins---lunch time---
• CM Features - 20 mins
• Dial Plan - 75 mins
• QOS - 22 mins
• Fax - 10 mins
• Misc - 15 mins
• ---360 mins (6 hr hrs)---
•
VoiceBootcamp
• Testing - Rest
326

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Truly Unified

CCIE Voice Lab Overview

• A 8-hour, hands-on lab exam.

• CCIE VOICE diagram

Voice Lab Sample Topology

For more updated Network Diagram please visit http://support.voicebootcamp.com

Distribution Edge IP Phone Tag

VoiceBootcamp Port must be

Cisco Devices use CDP protocol to discover all

Cisco 3550 or XL Switch

2 Cisco Catalyst 3500XL

Data and Voice VLAN in

3 Catalyst 3550 L3 Switch

Data and VOICE VLAN – Catalyst 3550 L3 Switch

• DNS configuration is required if name resolution is required

Server where DHCP will be hosted

• Choose System > DHCP > DHCP Server

DHCP Server IP phone request for IP

ip dhcp excluded-address 135.XX.67.1 135.XX.67.14

ip dhcp pool VOICE VLAN 10X VLAN 500

IOS DHCP Server

• Cisco router has the capability of becoming DHCP server

Toronto – Eastern Time Zone

SFO – Pacific Time Zone

NTP is often required by many network devices to provide a synchronized time

1 Call Setup CCM

• Phone 1 dials Phone 2

Device with Extension ‘3001’ Is

• This is second type of intra-cluster communication.

UCMPUB Device with Extension ‘3001’

Device with Extension ‘3001’

Select Service Activation

• Allows maximum of 500 region per Clusters

Add New IP Phones

• Directory Number is the extension number of this IP phone.

• Call Manager in an Cisco IOS router with special IOS.

Create the configuration files

cnf-file location flash:

phone basis or type

auto-reg-ephone - Can be used to prevent SCCP phone from registering automatically

Backup CME router telephony-service (2nd router)

Voice register pool  This command difine CME to support SIP

voice register dn 2 Define an extension

Voice register pool

mode cme  This command define CME to support SIP

• Phones in Cisco Unified CME

Directory Number ephone 1

An ephone is an Ethernet phone, and an ephone-dn is an Ethernet phone directory number. In CM

Directory Number (extension)

• SIP DN can also be shared line

• SIP based DN can be shared among multiple phone

• Provide BLF (Busy Lamp Field) notification for all the

Presence is defined using BLF feature of CME.

• “In shared line” call distribution to ring multiple phones at

Inbound call to 6011

09:00 06/500/05 6001 09:00 06/500/05

• Ephone-dn 5 is assigned to line 2 of both phone 1 and phone 2

• Forwards call to another DN if the intended DN does not

• Call-Forward Busy is used when line is in use

Inbound call to 6001 ephone-dn 11

• Create multiple ephone-dn entries with the same DN

• Only one phone rings at a time

• Channel huntstop works in a similar way for the

ephone-hunt allows CCME administrators to: