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CCIE VOICE Study Guide v3.0

Truly Unified

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Overview

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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 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.

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Courseagenda

333

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

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

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.

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Network Topology

CCIE VOICE

CCIE VOICE diagram Information Sheet containing DN, IP Address etc

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Voice Lab Sample Topology

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

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Chapter1

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Infrastructure and Services

ModuleOutline VLANS and VTP Server Configuring Cisco 6509 Catalyst Switches Configuring Cisco 35XX Catalyst Switches Configuring DHCP Servers. Configuring DHCP Relay Agent.

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VLAN

777

VLAN

SiSi

SiSi

EdgeSwitch

DistributionLayer

Phone VLAN = 101PC VLAN = 500

Desktop PC:

135.XX.166.0

IP Phone

135.XX.66.0

Port must be trunk if it is XL based Switch

IP Phone Tag packet with 802.1q for all voice traffic.Data traffic remain

untag

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.

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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

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

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CiscoDiscoveryProtocol

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CDP

Cisco Devices use CDP protocol to discover all devices are connected to its port.

Cisco 3550 or XL Switch

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

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Data and Voice VLAN in Catalyst 3500XL

PC VLAN = 500

Desktop PC 135.11.165.50

If it is a EtherSwitch and/or XL Switch, IP Phone port must be TRUNK and NATIVE vlan must

be set to data vlan

Catalyst3500XL

2 Cisco Catalyst 3500XL

Interface range FastEthernet0/1 - 4switchport mode trunkswitchport trunk encapsulation dot1qswitchport trunk native vlan 500switchport voice vlan 101spanning-tree portfast

Voice VLAN = 101

IP Phone 135.11.65.15Create VLAN

Switch# vlan dataSwitch(vlan) vlan 101 name RxVOICESwitch(vlan) vlan 500 name RxDATASwitch(vlan) vtp domain RACKXXSwitch(vlan) vtp server

Assign VLAN to Port

DataandVoiceVLANinCiscoCatalyst3500XL 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

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DataandVOICEVLAN

111111

Data and Voice VLAN in Catalyst 3550 L3

PC VLAN = 500

Desktop PC 135.11.165.50

802.1Q TrunkMake sure ROUTER PORT

Is trunk port with native vlan set to data vlan

Catalyst3550

3 Catalyst 3550 L3 Switch

interface FastEthernet2/0no ip addressswitchport access vlan 500switchport voice vlan 101spanning-tree portfast

IP Phone 135.11.65.15

Voice VLAN = 101

Create VLANSwitch# vlan dataSwitch(vlan) vlan 101 name RxVOICESwitch(vlan) vlan 500 name RxDATASwitch(vlan) vtp domain RACKXXSwitch(vlan) vtp server

Assign VLAN to Port

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.

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NetworkServicesNTP,DHCP,DNS

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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

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.

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UCM 7.0 DHCP

DynamicHostConfigurationProtocolDynamic 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

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DHCP Servers

Server where DHCP will be hosted

DNS Server

TFTP Server

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.

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DHCP Subnet

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.

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Networks Services: DHCP

PSTN

IP WAN

Toronto

SFO

CallManager

DHCP Server(135.7.100.20)

IP phone request for IPVia DHCP Broadcast

Interface vlan 101ip address 135.7.65.240ip helper-address 135.xx.100.11

DCHPServer 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.

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Networks Services: IOS DHCP

UK Office

Exclude IP address

VLAN 10X VLAN 500

VLAN 10X VLAN 500

Most IOS Router can act as a DHCP Server

ip dhcp excluded-address 135.XX.67.1 135.XX.67.14ip dhcp excluded-address 135.XX.67.51 135.xx.67.254ip dhcp pool VOICE

network 135.XX.67.0 255.255.255.0default-router 135.XX.67.240 option 150 ip 135.xx.67.240

!!interface fastEthernet0/0.10X (where X is Rack)Enacapsulation dot1q 10Xip address 135.xx.67.240 255.255.255.0!interface fastEthernet0/0.500Encapsulation dot1q 500 native vlanip address 135.XX.167.240 255.255.255.0

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

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NTP Configurations

Toronto Eastern Time Zoner7tor(config)#clock timezone EST -5r7tor(config)# ntp server

135.11.11.11SFO Pacific Time Zone

r7sfo(config)#clock timezone PST -8r7sfo(config)#ntp server 135.11.11.11

UK GMT 0r7uk(config)#clock timezone GMT 0r7sfo(config)#ntp server 135.11.11.11

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

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UCM NTP Server

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CallManager Basic

Unified Communication Manager 7.0

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).

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Deployment Models Centralized Call Processing

Toronto

SFOCallManagerCluster

AVVID ApplicationServer

UKCME Router

SRST

PSTN

IP backbone

CME

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.

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Simple Call Process

CCM

Unity 4.xVoice MailExchange 2K

Call Setup1

E.164 Lookup2Call Setup 3

Ring4

Off Hook5

Connect RTP

Stream6

Ring Back4

PSTN

IPWANSFOPhone 1

TOR Phone 2

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.

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CallManager Cluster &Redundancy

CallManager Group defines redundancy.

Group can have up to 3 CCM Server.

First server in the list is the Active CCM

PublisherStandby CCM

Primary CCM

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ccmpub

ccmsub

CCM GroupDefault

VCCluster

CallManagerClusterandRedundancy 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.

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CCM Device Registration

UCMPUB

UCMSUB-A

UCMSUB-B

Device with Extension 3001 Is Registered to Me (ccmpub)TCP Connect

(Active)SCCP

KeepAlive/30s

3001

CCM GROUP A1: ccmpub 2: ccmsubA3: ccmsubB

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.

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UCMPUB

UCMSUB-A

UCMSUB-B

SCCP KA

CCM Device Registration (conttd)

X3001

Device with Extension 3001 is UN-registered to Me (ccmpub)

Device with Extension 3001 is UN-registered to Me (CCM D)

Cisco CallManager List1: UCMPUB

2: UCMSUB-A3: UCMSUB-B

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.

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Tools Service Active

Select Service Activation

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.

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CallManager Server DNS-Less Environment

Enables Cisco IP Phones and other CCM-controlled devices to contact the CCM without resolving a DNS name

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.

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Call Manager Configuration ExampleDevice Registration and Redundancy

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

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Call Manager Configuration ExampleDefine Group to provide Redundancy

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.

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Time/Date

Group Define a name for the Time zone such as Eastern or New York EST etc. 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

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Region

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

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Device Pools

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

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Device Pools (contd)

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

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Enterprise Parameters

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

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Call Manager Configuration ExampleDevice Registration and Redundancy (contd)

Add New IP Phones

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.

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Device Level Configuration

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.

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IP Phone Line setting

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.

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Unified Communication Express 7.0

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CCME: Cisco Call Manager Express

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.

PSTN

IP WAN

Cisco Unified CME is an excellent choice for a single-site, standalone office. 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

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CME Setup

telephony-service load 7960-7940 P00308000500 max-ephones 100 max-dn 240ip source-address 135.Y.67.240 port 5000 ip qos dscp ef media ip qos dscp cs3 signal create cnf

Entering Telephony mode

Define Phone loads for upgrade/downgrade

Define max number of phone

Define what IP to bind CME to

QoS Settings for voice traffic

Create the configuration files

cnf-file location flash: cnf-file perphoneauto-reg-ephone

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

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UCME Redundant Router

PSTN

IP WAN

telephony-service ip source-address 135.7.67.240 port 5000 secondary 135.Y.67.241

voice-port 3/0/0 signal ground-start incoming alerting ring-only

telephony-service (2nd router)ip source-address 135.7.67.240 port 5000 secondary 135.Y.67.241

voice-port 3/0/0 signal ground-start incoming alerting ring-only ring number 3

Backup CME routerMust have Voice port Ring number set To higher then primary

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. NumberNumber 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.

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SIP: Setting Up Cisco Unified CME

Configure terminal

Voice register pool This command difine CME to support SIPmode cmesource-address ip-address 135.Y.67.240tftp-path http://www.voicebootcamp.com/filesmax-pool 25authenticate all realm voicebootcamp.com

voice register dn 2number 6001call-forward b2bua busy 6600huntstop channel 3!voice register pool 123busy-trigger-per-button 2id mac Y.Y.Y.Ytype 7961number 1 dn 2

Define an extension

Assign the extensionto a Phone

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)

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SCCP Setting UP CME for SCCP

telephony-service max-ephones 100 max-dn 240ip source-address 135.Y.67.240 port 5000

ephone-dn 2number 6001

ephone 1Mac-button 1:2

6001 60026003

ephone-dn 3 dual-linenumber 6002

ephone-dn 4 octo-linenumber 6003

ephone 2Mac-address Y.Y.Y.ybutton 1:3 2:4

Dual LineOcto-line

MAC address:Y.Y.Y.Y

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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

Ephone-Type Configuration

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Phone & Directory Number

Ethernet Phone or voice-register poolUsed by a Phone it selfEach phone must have a ephone X configure

Directory Number Number assign to line button on the phone

Single Line Dual Line Octo-Line

ephone-dn 2number 6001

ephone 1Mac-button 1:2

MAC address: X.X.X.X

6001 60026003

ephone-dn 3 dual-linenumber 6002

ephone-dn 4 octo-linenumber 6003

ephone 2Mac-address Y.Y.Y.ybutton 1:3 2:4

Single Line

Dual LineOcto-line

MAC address:Y.Y.Y.Y

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 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)

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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. Thats 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

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

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Line Comparison

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.

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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

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SIP None Shared-Line (Nonexclusive)

SIP DN can also be shared line

CME must be configured for SIP based network

voice register dn 2number 6001call-forward b2bua busy 6600huntstop channel 3!voice register pool 123busy-trigger-per-button 2id mac Y.Y.Y.Ytype 7961number 1 dn 2

6001

MAC address:Y.Y.Y.Y

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

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SIP Shared Line

voice register dn 2number 6001call-forward b2bua busy 6600shared-line max-calls 6huntstop channel 6!voice register pool 1busy-trigger-per-button 2id mac Y.Y.Y.Ytype 7961number 1 dn 2!voice register pool 2busy-trigger-per-button 3id mac X.X.X.Xtype 7965number 1 dn 2

6001

MAC address:Y.Y.Y.Y

6001

MAC address:X.X.X.X

Phone 1 Phone 2

busy-trigger-per-button

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

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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 managers 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

Presence is defined using BLF feature of CME.

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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.

CME - Shared Lines

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09:00 06/500/05 6001

60016011

Cisco CME

09:00 06/500/05 6002

60126011

Cisco CME

UK phone 1 UK phone 2

ephone-dn 10

number 6011 Shared DNephone 1

mac-address 2222.2222.2223

button 3:10

ephone 2

mac-address 2222.2222.2222

button 3:10

Inbound call to 6011

SCCP Shared Line

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

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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

Sequential Different DNs using Call Forward

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.

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CallManager Express Call Distribution/Hunting: Sequential Different DNs using Call Forward

09:00 06/10/07 6001

6001

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09:00 06/10/07 6002

6002

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IP phone 1

IP phone 2

ephone-dn 11

number 6001

call-forward busy 6002

Call-forward noan 6002 timeout 18

ephone 1

button 1:1

ephone-dn 2

number 6002

call-forward busy 6003

Call-forward noan 6003 timeout 18

ephone 2

button 1:2

Advice callmanagerexpress to forward calls to 6002 if 6001 is busy or does not answer after 18 seconds.

Inbound call to 6001

If phone 1 is busy or no answer, call is forwarded to 6002 in this case Phone 2

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

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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 usingpreference

[no] huntstop

huntstop channel

Only one phone rings at a time

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.

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09:00 06/500/05 6001

6001

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09:00 06/500/05 6001

6001

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IP phone 1

IP phone 2

ephone-dn 1

number 6001

no huntstop

preference 0

ephone 1

mac-address 3001.3001.3001

button 1:1

ephone-dn 2

number 6001

preference 1

ephone 2

mac-address 2222.2222.2222

button 1:2

Preference 0 is the highest priority and the default value, it does not appear in configuration

If DN is not available and there is a match and no hunt-stop configure the call will go to the next DN based on preference. For this work, both DN must have the same number.

Inbound call to 6001

If 6001 on phone 1 is busy, ring next match

CallManager Express Call Distribution: Sequential Same DN

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

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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

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.

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CCME Dual-line with Huntstop Channel

ephone-dn 1 dual-linenumber 6001no huntstophuntstop channel

ephone-dn 6 dual-linenumber 6001huntstop channelpreference 1

ephone 1mac-address 5001.5001.5001button 1:1 4:6

09:00 06/500/05 6001

60016001

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IP phone 1 Line 1 6001

Channel #1

Channel #2

Line 2 6001

Channel #1

Channel #2

Incoming Call to 6001

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

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CCME Dual-line without Huntstop Channel

ephone-dn 1 dual-linenumber 6001no huntstop

ephone-dn 6 dual-linenumber 6001preference 1

ephone 1mac-address 3001.3001.3001button 1:1 2:6

09:00 06/10/07 6001

60016001

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UK phone 16001

Line 1 6001

Channel #1

Channel #2

Line 2 6001

Channel #1

Channel #2

Incoming Call to 6001

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.

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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

There are three different kinds of ephone hunt groups.

Sequential ephone hunt groupsEphone-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 groupsThe 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 groupCalls 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.

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Ephone hunt

r5uk(config)#ephone-hunt 1 ?longest-idle longest idle huntingpeer peer huntingsequential sequential hunting

r5uk(config-ephone-hunt)#?EPHONE-HUNT configuration commands:auto enable automatic featuresdefault Set a command to its defaultsexit Exit from ephone hunt configuration modefinal final number for hunt grouplist list of number in hunt groupno Negate a command or set its defaultsno-reg not register pilot number to gatekeeperpilot pilot number for hunt grouppreference preference of pilot numberstatistics enable statistic information collecttimeout timeout in seconds for hunting

r5uk(config-ephone-hunt)#

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

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CCME Hunting

Ephone-hunt 1 seq

pilot 6500

list 6001, 6002

final 6000

timeout 5

ephone-hunt 2 peerpilot 6000list 6002, 6001, 6003final 3001 can not be 6500preference 1timeout 30no-reg

09:00 06/500/05 6001

6001

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09:00 06/500/05 6001

6001

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IP phone 1

IP phone 2

Inbound call to 6500

If 6001 is busy and/or not answering

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.

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CallManager ExpressDN 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.

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.

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CCME DN overlay Example ephone-dn 10number 6601no huntstoppreference 0

ephone-dn 11number 6601no huntstoppreference 1

Ephone-dn 12number 6601huntstoppreference 5

ephone 1mac-address 111.111.111button 1:1 2o10,11,12

ephone 1mac-address 111.111.112button 1:2 2o10,11,12

ephone 1mac-address 111.111.113button 1:2 2o12 11 10

06/500/05 6001

60016601

Cisco CME

UK phone 1

06/500/05 6002

60026601

Cisco CME

UK phone 2

06/500/05 6001

60026601

Cisco CME

UK phone 3

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.

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CallManager Express Shared DN overlay Example

ephone-dn 1number 6001

ephone 1mac-address 5001.5001.5001button 1:10,11,12

09:00 06/500/05 6001

60013001

Cisco CMEIP phone 1

09:00 06/500/05 6002

60023001

Cisco CMEIP phone 2

ephone-dn 2number 6002

ephone 2mac-address 2222.2222.2222button 1:2 2o1,11,12

ephone-dn 10number 3001

ephone-dn 11number 3002

ephone-dn 12number 3003

Overlay sets can be shared across multiple phones

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.

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Callmanager Express System Message

Allows you to change the default message on the IP Phone

telephony-servicesystem message Welcome to iNet?!

09:00 06/5/07 6001

6001

Welcome to iNet?!

Define a system messages such as company name or department name etc.

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CME Extension Mobility

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

Allow user to login to a physical other than their own phoneSales per going to remote branch office can login to one of the phone in BR office. Extension movies with the userUsually known as Follow Me NumberUser must login and logout to use EM FeaturesSome company use EM permanent solution to authenticate users

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

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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

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.

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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 1user name password passwordnumber 3002 type beep-ring speed-dial 2 5002 blfPin 1234

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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 1user name password passwordnumber 3002 type beep-ring speed-dial 2 5002 blfPin 1234

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Enabling an IP Phone for Extension Mobility

To enable the Extension Mobility feature on an individual Cisco Unified IP phone in Cisco Unified CME,

voice logout-profile 11user name password passwordnumber 3002 type beep-ring speed-dial 2 5002 blfPin 1234

Ephone 1mac-address Y.Y.Y.Ybutton 1:1type 7961logout-profile 11

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.

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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 5002 type beep-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

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.

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Configuring Transcoding in IOS

voice-card 1dsp services dspfarm

sccp local FastEthernet 0/1.101sccpsccp ccm 135.Y.67.240 identifier 1

sccp ccm group 123associate ccm 1 priorityassociate profile 1 register R1MTPkeepalive retries 5switchover method immediateswitchback method immediateswitchback interval 5

dspfarm profile 1 transcodecodec g711ulawcodec g711alawcodec g729ar8codec g719abr8maximum sessions 6associate application sccp

telephony-service ip source-address 10.5.49.500 port 5000 sdspfarm units 1 sdspfarm transcode sessions 40 sdspfarm tag 1 R1MTP

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 WWhheenn ddoo yyoouu nneeeedd TTrraannssccooddiinngg?? Ad hoc conferencingOne or more remote conferencing parties uses G.729. Call transfer and forwardOne 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 ExpressAn 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

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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.

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Presence Configurations

Enable Presence in CME

Configure terminalsip-uapresence

PresenceMax-subscriber 128Presence call-list

Enters SIP user-agent configuration mode to configure the user agent.

Allows the router to accept incoming presence requests

Enables presence service and enters presence configurationmode.

Globally enables BLF monitoring for directory numbers in call lists and directories on all locally registered phones

Enables presence service and enters presence configuration mode.

Enabling a Directory Number to be Watched

configure terminal ephone-dn 1 or voice register dn 1

number 6001allow-watch allow extenion to be watched

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.

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Presence on CME Speed Dial

Watcher can see the status of a internal as well as external numberUsing BLF Speed Dial to monitor the status of another extension

Ephone 1mac-address x.x.x.xbutton 1:1blf-speed-dial 1 6002 label Peter Smithpresence call-list

Voice register pool 1id mac-address x.x.x.xnumber 1 dn 1blf-speed-dial 1 6002 label Peter Smithpresence call-list

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

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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

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.

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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.

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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

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.

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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.

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Voice Gateways and Protocols

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Voice Gateway Protocols

H323 Gateway Other gateways/Trunk

SIP Trunk Gatekeeper Trunk

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

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Digital Voice Signaling: ISDN-PRI

PSTNE1 Framing

ISDN Q931ISDN Q921

isdn switch-type primary-ni!controller E1 0/0framing no-crc4linecode hdb3pri-group timeslots 1-24!int s0/0:15isdn incoming-voice voiceisdn switch-type primary-ni!voice-port 0/0:15cptone GB

!dial-peer voice 1 potsdestination-pattern 9.Tincoming called-number .direct-inward-dialport 0/0:15

Globally defines isdn switch type

D-channel (int s0/0:23) and voice-port will be automatically created once pri-group is defined on the T1 controller. D-channel carries the call information such as DNIS (called number) and ANI (calling number)

Defines T1-PRI under the T1 controller

Create pots dial-peer which defines voice call routing rules

ANI: Automatic Number Identification, a.k.a Calling number DNIS: Dialed Number Identification Service, a.k.a called number

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PSTN

CallManager

VoIP Signaling Protocols

H.323MGCP

Gatekeeper

VoIP Signaling:

SIP Gateway

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

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H.323 Gateway

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

Gateway status in CCM always remain Unknown

Framing

PRI Layer 3Layer 2

Cisco CallManager

PST

N H.225 and H.245 over TCP

PSTN IP

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.

Advantage of H323 Gateway

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

Disadvantage of h323 gateway No centralize management

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Basic H.323 IOS Configuration

controller T1 1/0framing esflinecode b8zspri-group timeslots 1-24!interface Serial1/0:23isdn switch-type primary-niisdn incoming-voice voice!dial-peer voice 1 voipdestination-pattern 3...session target ipv4:135.XX.100.12codec g711ulawdtmf-relay h245-alphanumeric!dial-peer voice 9 potsdestination-pattern 9Tdirect-inward-dialincoming called-number .Tport 1/0:23

Defines T1-PRI as PSTN signaling

Dial Peer for VoIP Leg

D-channel and its configurations

Pots dial-peer pointing to the PRI with destination-pattern, pots peers strips explicitly matched digit(s) in destination-pattern

Destination-pattern for digit matching

Session target pointing to ipaddress of remote H.323 peer: i.e. Call Managers IP addr.

Use g711u codec. Default is g729

Enables DTMF relay using H245-alpha. Default is disabled

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

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Additional H.323 IOS Configuration Options

interface loopback 0ip address XX.33.33.33 255.255.255.0h323-gateway voip interfaceh323-gateway voip bind srcaddr XX.33.33.33!voice class h323 1h225 timeout setup 5!voice class codec 1codec preference 1 g729r8codec preference 2 g711ulaw!dial-peer voice 1 voipdestination-pattern 3...session target ipv4:135.XX.100.12voice-class h323 1voice-class codec 1!dial-peer voice 2 voipdestination-pattern 3...session target ipv4:135.XX.100.11voice-class h323 1voice-class codec 1preference 1

Forces this gateway to use the loopback interface for all H.323 signal and UK traffic.

H.225 setup redundancy: try a second voip dial-peer if the remote H.323 peer does not response in 5 seconds.

H.245 codec negotiation flexibility: negotiate to g729 if possible; otherwise g711ulaw is okay too.

Try this dial-peer first if 3 is match because it has the highest preference: 0. Default preference value, therefore invisible in dial-peer configuration.

If the IP host in dial-peer 1 (135.XX.100.12) does not response H.225 setup in 5 seconds, try this dial-peer as it has lower preference.

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

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Call Manager H.323 Gateway Configuration

1

2a

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

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Call Manager H.323 Gateway Config. (contd)

2b Continued from CCM H.323 Gateway Configuration Page:

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.

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H323 Dial Peer

7 Digit (with 9 access code)

dial-peer voice 7 pots

destination-pattern 9[2-9]

forward-digits 7

port 1/0:23

11 (long distance)

dial-peer voice 11 pots

destination-pattern 91[2-9]..[2-9]

forward-digits 11

port 1/0:23

911 callsDial-peer voice 911 pots

Destination-pattern 911

Forward-digits 3

Port 1/0:23

Overseas or internationalDial-peer voice 111 pots

Destination-pattern 9011T

Port 1/0:23

Prefix 011

Any explicit match will be discarded dial-peer voice 11 pots 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.

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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

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.

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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.

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IOS MGCP PRI Backhaul Configurationhostname rXsfo!mgcpmgcp call-agent 135.XX.100.11mgcp bind control source looopbac0mgcp bind media source loopback0!ccm-manager redundant-host 135.XX.100.12ccm-manager mgcpccm-manager fallback!

controller T1 1/0linecode b8zsframing esfpri-group timeslots 1-7 service mgcp!interface Serial1/0:23no ip addressno logging event link-status isdn incoming-voice voiceisdn bind-l3 ccm-manager !dial-peer voice 101 potsservice mgcpappport 1/0:23

Must match Domain Name on MGCP Gateway page on CCM

Defines Primary Call-agent: the ipaddress of primary CCM

Enables MGCP process globally

Defines secondary call-agent

Defines on the T1 controller that the PRI ports will be serviced by MGCP

Defines MGCP as the call application under pots dial-peer

Under D-channel, binds L3 (Q.931) to call manager

MGCP version 0.1 with CallManager

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

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MGCP: Call Manager Configuration

1

2Must match with hostname and ip domain-name (if applicable) on the IOS gateway

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

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MGCP: Call Manager Configuration (contd)

3

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.

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Useful IOS MGCP Verification Commands

GW1#sh isdn statGlobal ISDN Switchtype = primary-niISDN Serial1/0:23 interface

dsl 0, interface ISDN Switchtype = primary-niL2 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 = 0The Free Channel Mask: 0x8000003FNumber of L2 Discards = 2, L2 Session ID = 30Total Allocated ISDN CCBs = 0

When you type show isdn status in MGCP router, Layer 2 Status will be multiple frame established only when CCM is registered.

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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:@.

A signaling interface (trunk) must be configured to receive/send calls.

A SIP network uses the following components: SIP Proxy ServerThe 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 ServerThe 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 ServerThe 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.

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A unique SIP address that appears similar to an e-mail address and uses the format sip:@. 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.

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SIP Gateway (contd)

SIP signaling interfaces connect Cisco CallManagernetworks 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

SIP and CallManager Connectivity All protocols require that either a signaling interface (trunk) or a gateway