Truly Unified

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CCIE VOICE Study Guide v3.0

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 preconfigured.

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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.

Courseagenda

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

Day 4

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Section 10 Integrating with Unity Express 7.0 Section 11 integrating with Unified Contact Center Express/ Advanced Scripting Section 13 UC Application - IPMA, EM , Mobility, Single Number Reach, Mobile Access Section 14 QoS 8 hours Lab simulation

Section 09 Integration with Unity Connection 7.0, Advanced Unity Connection Configuration

Section 12 Integration with Cisco Unified Presence, Advanced Unified Presence & Microsoft OCS integration

Day 5

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

CCIE VOICE diagram Information Sheet containing DN, IP Address etc

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NetworkTopology

Voice Lab Sample Topology

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

Chapter1

Infrastructure and Services

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ModuleOutline
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 Layer

Edge Switch

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

PC VLAN = 500

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Port must be trunk if it is XL based Switch

Phone VLAN = 101

IP Phone

Desktop PC:
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135.XX.66.0

135.XX.166.0

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

CiscoDiscoveryProtocol

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

Create VLAN Switch# vlan Switch(vlan) Switch(vlan) Switch(vlan) Switch(vlan)

IP Phone 135.11.65.15

Desktop PC 135.11.165.50

data If it is a EtherSwitch and/or XL vlan 101 name RxVOICE Switch, IP Phone port must be vlan 500 name RxDATA TRUNK and NATIVE vlan must vtp domain RACKXX be set to data 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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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

DataandVOICEVLAN

Data and Voice VLAN in Catalyst 3550 L3

3 Catalyst 3550 L3 Switch
Voice VLAN = 101 PC VLAN = 500

Catalyst 3550

Create VLAN 802.1Q Trunk Switch# vlan data Make sure ROUTER PORT Switch(vlan) vlan 101 name RxVOICE Is trunk port with native vlan Switch(vlan) vlan 500 name RxDATA set to data vlan Switch(vlan) vtp domain RACKXX Switch(vlan) vtp server interface FastEthernet2/0 no ip address switchport access vlan 500 Assign VLAN to switchport voice vlan 101 Port spanning-tree portfast

IP Phone 135.11.65.15

Desktop PC 135.11.165.50

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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.

NetworkServicesNTP,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

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

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DynamicHostConfigurationProtocol
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

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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 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.

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.

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. displays the item that you choose.

Note : To reverse the sort order, click the up or down arrow, if available, in the list header. The window

Networks Services: DHCP

DHCP Server (135.7.100.20) IP phone request for IP Via DHCP Broadcast

PSTN
CallManager

IP WAN
Toronto

SFO

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

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 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) Enacapsulation dot1q 10X ip address 135.xx.67.240 255.255.255.0 ! interface fastEthernet0/0.500 Encapsulation dot1q 500 native vlan ip address 135.XX.167.240 255.255.255.0
VLAN 10X VLAN 500

VLAN 10X

VLAN 500

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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

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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Unified Communication Manager 7.0

CallManager Basic

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
AVVID Application Server

Centralized Call Processing

SRST

PSTN
CallManager Cluster

SFO

IP backbone
CME

Toronto

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

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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
SFO Phone 1 Connect 6 RTP Stream

CCM
2 E.164 Lookup

4
Ring Back Call Setup

IPWAN

3 4
Ring

PSTN

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 Cluster CCM Group Default
Publisher Standby CCM

ccmpub

Primary CCM

ccmsub

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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 Registered to Me (ccmpub)

TCP Connect (Active) SCCP KeepAlive/30s

UCMPUB

3001

UCMSUB-A

UCMSUB-B
CCM GROUP A 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 (conttd)

UCMPUB
SCCP KA

3001

UCMSUB-A

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

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

UCMSUB-B
Cisco CallManager List 1: UCMPUB 2: UCMSUB-A 3: 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

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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

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 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

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Call Manager Configuration Example

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

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

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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 used per audio call. include overhead.

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The audio codec determines the type of compression and the maximum amount of bandwidth that is The video call bandwidth comprises the sum of the audio bandwidth and video bandwidth but does not 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 (contd)

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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

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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 Example

Device Registration and Redundancy (contd)

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

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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.

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 240 ip 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 perphone auto-reg-ephone

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IP source-address defines what IP address you want the Callmanager Express to bind to.

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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

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

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

PSTN

IP WAN

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

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

Assign the extension to a Phone

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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: tftppath 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 max-ephones 100 max-dn 240 ip source-address 135.Y.67.240 port 5000
ephone-dn 2 number 6001 ephone 1 Macbutton 1:2 ephone-dn 3 dual-line number 6002 ephone-dn 4 octo-line number 6003 ephone 2 Mac-address Y.Y.Y.y button 1:3 2:4

Dual Line Octo-line 6001 6002 6003 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 pool Used by a Phone it self Each phone must have a ephone X configure Directory Number Number assign to line button on the phone Single Line Dual Line Octo-Line Single Line ephone-dn 2 number 6001 ephone 1 Macbutton 1:2 ephone-dn 3 dual-line number 6002 ephone-dn 4 octo-line number 6003 ephone 2 Mac-address Y.Y.Y.y button 1:3 2:4

Dual Line Octo-line 6001 6002 6003 MAC address: Y.Y.Y.Y

MAC address: X.X.X.X

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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. 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

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r1uk(config)#ephone-dn 1 r1uk(config-ephone-dn)#number 6001

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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.

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

6001 MAC address: Y.Y.Y.Y

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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

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

Phone 1 6001

Phone 2 6001 MAC address: Y.Y.Y.Y

MAC address: X.X.X.X

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In this scenario first two calls will arrive on Phone 1 and 3rd call will arrive on Phone 2 because of busytrigger-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 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

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Presence is defined using BLF feature of CME.

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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.

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SCCP Shared Line

Inbound call to 6011

09:00 06/500/05

6001

6001 UK phone 1 Cisco CME ephone-dn 10 number 6011 Shared DN ephone 1 mac-address 2222.2222.2223 button 3:10 6011 UK phone 2

09:00 06/500/05 6002

6012 6011 Cisco CME

ephone 2 mac-address 2222.2222.2222 button 3:10

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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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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
09:00 06/10/07 6001

ephone-dn 11 number 6001 call-forward busy 6002 Call-forward noan 6002 timeout 18 ephone 1 button 1:1

6001 IP phone 1
If phone 1 is busy or no answer, call is forwarded to 6002 in this case Phone 2

VoiceBootcamp Inc Advice callmanager express to forward calls to 6002 if 6001 is busy or does not answer after 18 seconds.

ephone-dn 2 number 6002 call-forward busy 6003 Call-forward noan 6003 timeout 18 ephone 2 button 1:2

09:00 06/10/07

6002

6002 IP phone 2 VoiceBootcamp Inc.

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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

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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
Inbound call to 6001
09:00 06/500/05 6001

ephone-dn 1 number 6001 no huntstop preference 0 ephone 1

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

6001 IP phone 1
If 6001 on phone 1 is busy, ring next match

VoiceBootcamp Inc. If DN is not available and there is a match and no huntstop configure the call will go to the next DN based on preference. For this work, both DN must have the same number.
09:00 06/500/05 6001

mac-address 3001.3001.3001 button 1:1

ephone-dn 2 number 6001 preference 1 ephone 2

6001 IP phone 2 VoiceBootcamp Inc.

mac-address 2222.2222.2222 button 1:2

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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 ephonedn 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

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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 6001

IP phone 1 VoiceBootcamp Inc.

ephone-dn 1 dual-line number 6001 no huntstop huntstop channel ephone-dn 6 dual-line number 6001 huntstop channel preference 1 ephone 1

Line 1

6001
Channel #1 Channel #2

Line 2

6001
Channel #1 Channel #2
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button 1:1 4:6

mac-address 5001.5001.5001

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

6001 6001 UK phone 1 6001 VoiceBootcamp Inc.

ephone-dn 1 dual-line number 6001 no huntstop ephone-dn 6 dual-line number 6001 preference 1 ephone 1 mac-address 3001.3001.3001 button 1:1 2:6

Incoming Call to 6001 Line 1 6001

Channel #1 Channel #2

Line 2

6001
Channel #1 Channel #2

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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

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There are three different kinds of ephone hunt groups.

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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.

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)#

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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
Ephone-hunt 1 seq pilot 6500 list 6001, 6002 final 6000 timeout 5
ephone-hunt 2 peer pilot 6000 list 6002, 6001, 6003 final 3001 can not be 6500 preference 1 timeout 30 no-reg 09:00 06/500/05 6001

Inbound call to 6500

6001 IP phone 1 VoiceBootcamp Inc.

If 6001 is busy and/or not answering

09:00 06/500/05 6001

6001

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IP phone 2

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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.

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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

06/500/05 6001

6001 6601 UK phone 1 Cisco CME

06/500/05 6002

ephone-dn 10 number 6601 no huntstop preference 0 ephone-dn 11 number 6601 no huntstop preference 1

6002 6601 UK phone 2 Cisco CME

06/500/05 6001

Ephone-dn 12 number 6601 huntstop preference 5 ephone 1 mac-address 111.111.111 button 1:1 2o10,11,12 ephone 1 mac-address 111.111.112 button 1:2 2o10,11,12 ephone 1 mac-address 111.111.113 button 1:2 2o12 11 10

6002 6601 UK phone 3 Cisco CME

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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 ephonedns, 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 6001 09:00 06/500/05 6002

6001 IP phone 1 Cisco CME ephone-dn 1 number 6001 ephone 1 mac-address 5001.5001.5001 button 1:10,11,12 ephone-dn 10 number 3001 ephone-dn 11 number 3002 ephone-dn 12 number 3003 3001 IP phone 2 Cisco CME ephone-dn 2 number 6002

6002 3001

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

Overlay sets can be shared across multiple phones

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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?!

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

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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

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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
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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Router(config-telephony)# url authentication http://192.0.2.0/CCMCIP/authenticate.asp secretname psswrd

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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 1 user name password password number 3002 type beep-ring speed-dial 2 5002 blf Pin 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 1 user name password password number 3002 type beep-ring speed-dial 2 5002 blf Pin 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 11 user name password password number 3002 type beep-ring speed-dial 2 5002 blf Pin 1234 Ephone 1 mac-address Y.Y.Y.Y button 1:1 type 7961 logout-profile 11

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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 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

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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 dsp services dspfarm sccp local FastEthernet 0/1.101 sccp sccp ccm 135.Y.67.240 identifier 1 sccp ccm group 123 associate ccm 1 priority associate profile 1 register R1MTP keepalive retries 5 switchover method immediate switchback method immediate switchback interval 5 dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g719abr8 maximum sessions 6 associate application sccp telephony-service ip source-address 10.5.49.500 port 5000 sdspfarm units 1 sdspfarm transcode sessions 40 sdspfarm tag 1 R1MTP

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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

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When do you need Transcoding? 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

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 speeddial buttons and directory call lists for missed calls, placed calls, and received calls.

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Presence Configurations
Enable Presence in CME Configure terminal sip-ua presence Presence Max-subscriber 128 Presence 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 configuration mode. Enables presence service and enters presence configuration mode. Enabling a Directory Number to be Watched Globally enables BLF monitoring for directory numbers in call lists and directories on all locally registered phones

configure terminal ephone-dn 1 or voice register dn 1 number 6001 allow-watch allow extenion to be watched

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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

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Blf-speed-dial is a special speed dial that can track the status of the destination device.

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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

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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

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 cfwdnoan 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

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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

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 cfwdnoan 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

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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 E1 Framing

PSTN
Globally defines isdn switch type

isdn switch-type primary-ni ! controller E1 0/0 framing no-crc4 linecode hdb3 pri-group timeslots 1-24 ! int s0/0:15 isdn incoming-voice voice isdn switch-type primary-ni ! voice-port 0/0:15 cptone GB ! dial-peer voice 1 pots destination-pattern 9.T incoming called-number . direct-inward-dial port 0/0:15

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

VoiceBootcamp

Create pots dial-peer which defines voice call routing rules

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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 SIP Gateway

VoiceBootcamp
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.

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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 Layer 2 Framing

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

PSTN

H.225 and H.245 over TCP

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 Gateway status in CCM always remain Unknown

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Basic H.323 IOS Configuration

Defines T1-PRI as PSTN signaling controller T1 1/0 framing esf linecode b8zs pri-group timeslots 1-24 ! interface Serial1/0:23 isdn switch-type primary-ni isdn incoming-voice voice ! dial-peer voice 1 voip destination-pattern 3... session target ipv4:135.XX.100.12 codec g711ulaw dtmf-relay h245-alphanumeric ! dial-peer voice 9 pots destination-pattern 9T direct-inward-dial incoming called-number .T port 1/0:23 D-channel and its configurations Dial Peer for VoIP Leg

Destination-pattern for digit matching Session target pointing to ip address of remote H.323 peer: i.e. Call Managers IP addr. Use g711u codec. Default is g729 Enables DTMF relay using H245alpha. Default is disabled Pots dial-peer pointing to the PRI with destination-pattern, pots peers strips explicitly matched digit(s) in destination-pattern
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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
ip address XX.33.33.33 255.255.255.0 h323-gateway voip interface h323-gateway voip bind srcaddr XX.33.33.33

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.

! voice class h323 1 h225 timeout setup 5 ! voice class codec 1 codec preference 1 g729r8 codec preference 2 g711ulaw ! dial-peer voice 1 voip destination-pattern 3... session target ipv4:135.XX.100.12 voice-class h323 1 voice-class codec 1 ! dial-peer voice 2 voip destination-pattern 3... session target ipv4:135.XX.100.11 voice-class h323 1 voice-class codec 1 preference 1

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

1

2a

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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. (contd)

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

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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) 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 calls
Dial-peer voice 911 pots Destination-pattern 911 Forward-digits 3 Port 1/0:23

Overseas or international
Dial-peer voice 111 pots Destination-pattern 9011T Port 1/0:23 Prefix 011

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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.

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

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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 ! mgcp mgcp call-agent 135.XX.100.11 mgcp bind control source looopbac0 mgcp bind media source loopback0 ! ccm-manager redundant-host 135.XX.100.12 ccm-manager mgcp ccm-manager fallback! controller T1 1/0 linecode b8zs framing esf pri-group timeslots 1-7 service mgcp ! interface Serial1/0:23 no ip address no logging event link-status isdn incoming-voice voice isdn bind-l3 ccm-manager ! dial-peer voice 101 pots service mgcpapp port 1/0:23 Must match Domain Name on MGCP Gateway page on CCM Enables MGCP process globally Defines Primary Call-agent: the ip address of primary CCM Defines secondary call-agent MGCP version 0.1 with CallManager Defines on the T1 controller that the PRI ports will be serviced by MGCP Under D-channel, binds L3 (Q.931) to call manager Defines MGCP as the call application under pots dial-peer
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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

1

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

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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 (contd)

3

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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

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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.

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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.

 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 (contd)

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
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

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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 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.

VoiceBootcamp

SIP Gateway Call

PSTN

IP

Cisco CallManager

IP

MTP

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Forwarding a DTMF Calls

94168391717

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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-connections h323 to sip Dial-peer voice 1 pots application sessions destination-pattern 9T incoming called-number . direct-inward-dial port 1/0:23 dial-peer voice 101 voip destination-pattern [35] session protocol sipv2 session target ipv4:135.11.100.11 dtmf-relay sip-notify codec g711ulaw Allow PSTN calls (h323) to reach SIP network Application session defines that standard session application will be invoked for this dial peer

Define a VoIP Dial-peer to send calls to CCM Must change the protocol to SIPV2 as default is h323 For Cisco IP phone to work, you must use SIPNOTIFY As DTMF Relay

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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

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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

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

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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

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Translation contd
Example 2 This example shows how to replace any occurrence of "123" at the start of a number 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 rXuk#test voice translation-rule 1 1234 Matched with rule 1 Original number: 1234 Translated number: 4564

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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

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.

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Translation Contd
Wildcard Definition

. 0 to 9,*,# [0-9] * +

Any single digit Any specific character Any range or sequence of characters Modifiermatch none or more occurrences Modifiermatch one or more occurrences

VoiceBootcamp
Various wildcard can be used to construct your match pattern.

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Translation Contd
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

.+ ^$

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Some example of wildcard usages

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Translation Contd
Example 2 This example replaces all numbers with "5554000". voice translation-rule 2 rule 1 /.*/ /5554000/ rXuk#test voice translation-rule 2 123 Matched with rule 1 Original number: 123 Translated number: 5554000

Replace all numbers with 5554000 Useful for changing caller ID for a company to a specific number

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Translation Contd
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 Original number: 123 Translated number: 5554000 router#test voice translation-rule 2 "" Didn't match with any of rules

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Replace all number except empty string.

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ISDN D Channel
R1sfo(config) interface serial 1/0:23 r1sfo(config-if)#isdn ? bchan-number-order bind-l3 caller calling-number outgoing outgoing-voice overlap-receiving Specify bchannel starting number order Bind Layer 3 protocol to signaling interface Specify incoming telephone number to be verified Specify Calling Number included for outgoing calls Options for outgoing IEs and messages Specify information transfer capability for voice Specify if the interface will do Overlap Receiving calls

bchan-number-order used this to change the channel selection order either top down or bottom 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

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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
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

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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.

RAS Gatekeeper Registration Illustrated

Gatekeeper

RRQ
Hello: I am Registering My Name or E.164 Address (GW-A - PREFIX: 514)

RRQ RCF RCF

Hello: I am Registering My Name or E.164 Address (Gateway B - Prefix 416)

GW A

IP QoS WAN
UDP Transport Port 1719 RRQRegistration Request RRJRegistration Reject RCFRegistration Confirm

GW B

VoiceBootcamp
Address TranslationTranslates 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.

RASRegistration Admission and Status

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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)
ARQ (Admission Request): I Have a Call for 416-839-1717 Dynamic Table
GW-B Prefix 416 IP: 1.1.1.1

ARQ ACF

IP QoS WAN

GW-A Prefix: 514 IP: 2.2.2.2

H.323 Call Set-Up GW-a GW-B

VoiceBootcamp

ACF (Admission Confirm): Yes You Can, Use GW-B IP Address 1.1.1.1

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Admission ControlControls 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 ControlConsists 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 ManagementThe 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 Gatekeeper A
GK

QoS WAN

H.323 Gatekeeper B
GK

Gatekeeper Zone A
H.323 GW

Gatekeeper Zone B

Local PSTN

Local PSTN

Local PSTN

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.

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Gatekeeper Inter-zone Communication:

GKA
LRQ LCF ACF ACF

GKB

IP Network
ARQ H.225 Fast Start H.225 Fast Connect UK ARQ

Phone A 3001

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Gateway A Gateway B

Phone B 5001
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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 NetworkGateways Only Small NetworkSimplified with a Gatekeeper

VoiceBootcamp
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.

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Cisco IOS Gatekeeper: Common Terms

Zonea collection of nodes for routing calls (can be H.323 clients, CallManager clusters, or H.323 Gateways) Zone Prefixa string of numbers used associate a dialed number to a zone Tech Prefixa unique string used to group endpoints of the same type together Default TechnologyGateways that register with a tech prefix that are used for default routing of any E.164 address that is otherwise unresolved

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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

Call to 14163133001

1 10 11
Local Zone: SFO Zone Prefix: 1408*

GK = Gatekeeper

2 GK

Local Zone: TOR Zone Prefix: 1416* TORRTR Technology Prefix: 1# TOR Ph-1 14163133001

SFO Phone 1 14086165001

SFORTR Technology Prefix: 1#

Default Technology Prefix: 1#

1 2 3 4 5

ARQ to 14163133001

14163133001 registered? No

6 7 8 9 10
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Technolgy prefix match? No

Was a technology prefix found? No

Zone prefix match? Yes

Default technology prefix set? 1#

VoiceBootcamp
Is TOR a local zone? Yes ACF, destination TORRTR

Target zone = TOR

Select a gateway in TOR with technology prefix 1#.

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.

Gatekeeper Call Routing: Zone Prefixes & Technology Prefixes

Call to 14163133001

1 10 11
Local Zone: SFO Zone Prefix: 1408*

GK = Gatekeeper

2 GK

Local Zone: TOR Zone Prefix: 1416* TOR Technology Prefix: 1#

Is TOR local? Yes

SFO Phone 1 14086165001

1 2 3 4 5

SFO Technology Prefix: 1# Dial Peer Technology Prefix: 1#

ARQ to 1#14163133001

TOR Ph1 14163133001

6 7 8 9 10
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Technolgy prefix match? Yes, 1#

14163133001 registered? No

Hopoff prefix? No

Was a technology prefix found? Yes

VoiceBootcamp
Target zone = TOR ACF, destination TOR

Zone prefix match? Yes

Select a gateway in TOR with tech prefix 1#.

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.

Gatekeeper Call Routing: Zone Prefixes & Registered Numbers

Call to 14163133001

1 7 8
Local Zone: SFO Zone Prefix: 1408*

GK = Gatekeeper

2 GK

Local Zone: TOR Zone Prefix: 1416* TORRTR TOR Ph 1 Technology Prefix: 1# 14163133001 E.164 14163133001

SFO Phone 14086165001

SFORTR Technology Prefix: 1#

1 2 3 4 5

ARQ to 14163133001 Technology prefix match? No Zone prefix match? Yes Target zone = TOR Is TOR a local zone? Yes

14163133001registered? Yes ACF, destination TORRTR

6 7

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.

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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
<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*)

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<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 9 trunk_1 zone prefix VOICERACKXX 3... gw-priority 0 UKGW zone prefix VOICERACKXX 5... gw-priority 10 trunk_2 zone prefix VOICERACKXX 5... gw-priority 9 trunk_1 zone prefix VOICERACKXX 5... gw-priority 0 UKGW zone prefix VOICERACKXX 6... gw-priority 0 trunk_2 zone prefix VOICERACKXX 6... gw-priority 0 trunk_1 zone prefix VOICERACKXX 6... gw-priority 10 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

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. zone prefix VOICERACKXX 3... gw-priority 10 trunk_2

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zone prefix VOICERACKXX 3... gw-priority 9 trunk_1 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. zone prefix VOICERACKXX 3... gw-priority 0 UKGW 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 Define a h323 alias Register to GK XX.11.11.11 with zone RACKXX Disable gateway functions Enable Gateway to register to GK

H323-gateway voip interface H323-gateway voip h323-id UKGW H323-gateway voip id VOICERACKXX ipaddr XX.11.11.11 H323-gateway voip tech-prefix 1#

No gateway Gateway Dial-peer voice 3000 voip

Destination-pattern [3,5] Session target ras Tech-prefix 1#
Dialpeer sends call to RAS which is gatekeeper

Num-exp should be used to strip the Tech-prefix

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When use rdial 3001 or 5001 this dial peer will add 1#. So gatekeeper sees the incoming call as 1#3001 Based on the 1# and zone prefix, GK will route the call accordingly.

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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

1

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To route calls via Gatekeeper, you must add gatekeeper and trunk.

121

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 (contd)

3

Translation pattern must be used To remove tech-prefix

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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*

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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.

Debug gatekeeper main 10 or 5 is a hidden command and provide detail information such as why the call failed?

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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
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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 bandwidth total zone VOICERACK66 512 bandwidth intrazone VOICERACK66 64 bandwidth session zone VOICERACK66 16 Specifies the default value for all zones Specifies the total amount of bandwidth for H.323 traffic allowed in the zone Specifies the total amount of bandwidth for H.323 traffic from the zone to any other zone. Specifies the maximum bandwidth allowed for a session in the zone.

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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

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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
Route Pattern

The IP Routing of IP Telephony

9.1416XXXXXXX
CallManager Head office Gatekeeper
GK

Cluster

5000

IP WAN

Router/GW

PSTN 914163133001

3001

CallManager Routes Two Basic Call Types:

On-Net CallsDestination Directory Number (DN) is registered with CallManager Off-Net Calls External route patterns must be configured on CallManager

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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

User Dials 6500

Route Patterns

6XXX 62XX

User Dials 6234

Directory Numbers

6234
6234

CallManager matches the most specific pattern (longestmatch logic) An IP phone directory number is a special case of route pattern that matches a single number

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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

Route Pattern
Matches dialed number for external calls Performs digit manipulation (optional) In IOS we call this dial-peer

External Route Elements in CallManager

Route Pattern

Route List
Chooses path for call routing Points to prioritized route groups
1st Choice

Route List

2nd Choice

Route Group Route Group

Choose the right devices.
GK IP WAN

Route Group
1st Choice 2nd Choice

Devices
Gateways (H.323, MGCP) Gatekeeper Inter-Cluster Trunk (remote CM)

VoiceBootcamp

6608 t1 & 1 fxo

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.)

Configuration Order
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Route Group

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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

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Route List

To use Local Route group Route List must contain this RG

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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.

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Route Pattern

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Select the Route List from the list
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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

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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 A Phone C

Partition = A NO ONE CAN REACH EACH OTHER

Partition = C

Partition = A

Phone D

Parition blocks inbound communication unless calling party has CSS with Called Partys partition in it
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Phone B

Partition = B

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.

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CSS
Partition = A A Partition = C CSS_C Partition C Partition B

CSS_A Partition A Partition B

Phone A Phone B Phone C Phone D

B

B, D but not C A, C but not D D only C, A and B

CSS_B Partition A Partition C

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Partition = A Partition = B

CSS_D Partition C Partition A

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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
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.

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Partition and CSS Example Local calls

DD - Pre-dot Prefix 1408 CSS_TOR_LOCAL PT_TOR_911 PT_TOR_LOCAL

TOR 6608 T1 TOR-A 9.[2-9]XXXXXX

PT_TOR_LOCAL

RG_TOR

TOR

8391717
DD - Pre-dot Prefix N/A

RL_TOR_LOCAL

9.8391717

TOR-S

9.6391717 9.[2-9]XXXXXX SFO PT_SFO_LOCAL

CSS_SFO_LOCAL PT_SFO_911 PT_SFO_LOCAL

RL_SFO_LOCAL RG_SFO

8391717 SFO MGCP

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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

Partition and CSS Example LD To North America

CSS_TOR_LOCAL PT_TOR_911 PT_TOR_LOCAL PT_TOR_LD

TOR 6608 T1
RG_TOR

9.1[2-9]XX[2-9]XXXXXX
PT_TOR_LD

TOR-A

DD - Pre-dot Prefix N/A

TOR

RL_TOR_LD
DD - Pre-dot Prefix N/A

9.14088391717 TOR-S

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
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RG_SFO

SFO

SFO MGCP

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.

Partition and CSS Example

LD calls from TOR to SFO only use SFO GW

14088391717
CSS_TOR_LOCAL PT_TOR_911 PT_TOR_LOCAL PT_TOR_LD PT_TOR_LD_SFO

TOR 6608 T1
RG_TOR

9.1408[2-9]XXXXXX 91408.[2-9]XXXXXX 9.1[2-9]XX[2-9]XXXXXX

PT_TOR_SFO_LD

TOR-S

DD - Pre-dot Prefix 1408

TOR

RL_TOR_LD_SFO
DD - Pre-dot Prefix N/A

9.14088391717 TOR-A

RG_SFO

SFO SFO MGCP

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.
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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.

Partition and CSS Example

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 Gatekeeper CSS_TOR_LOCAL
PT_TOR_911 PT_TOR_LOCAL PT_TOR_LD PT_TOR_TOLL

RG_GK

8.6XXX
PT_TOR_TOLL

TOR-A
RL_TOR_TOLL

DD - Pre-dot Prefix 1#

IP WAN

86001

TOR-S

DD - Pre-dot Prefix 0114402896

PSTN UK

RG_TOR

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.
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TOR 6608 T1

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 PreDot. 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

CSS_TOR_LOCAL PT_TOR_911 PT_TOR_LOCAL

Device Pool RG_TOR

DD - Pre-dot Prefix 1408

TOR 6608 T1
RG_TOR

TOR

9.[2-9]XXXXXX
PT_TOR_LOCAL

4
DD - Pre-dot Prefix N/A

8391717

9.8391717 1
RL_PSTN

Local Route Group

9.6391717

3
RG_SFO

CSS_SFO_LOCAL PT_SFO_911 PT_SFO_LOCAL

VoiceBootcamp
Device Pool RG_SFO SFO MGCP

9.[2-9]XXXXXX SFO PT_SFO_LOCAL

8391717

DD - Pre-dot Prefix N/A => Prefix 1416

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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 devices (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 to PSTN (e.g. 9.! or 9.@), scroll to Calling Party Transformations Check the Use Calling Party's External Phone Number Mask option

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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 partys Phones 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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Digit Prefix can be configured on both Calling Number as well as Called Number

Calling Party Transformation Order

1. Apply the external phone number mask 2. Apply the calling party transformation mask 3. Apply prefix digits
Directory Number External Phone Number Mask Calling-Party Transformation Mask Caller ID
41685XX000

45062 41671XXXXX 4167145062 51485XX000 5148545000

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ThecallingpartynumberassociatedwithacallroutedthroughUnifiedCMmightsometimeshavetobeadapted beforeitispresentedtoaphoneortothePSTN.Callsofferedtogatewaysmightrequirethecallingpartynumber bemanipulatedtoadaptittotherequirementsofthetelephonycarriertowhichthegatewayisconnected.For example,acallfrom+14167254000offeredtoagatewaylocatedinFrancemighthavetorepresentthecalling numberas0014167254000,withaCallingPartyNumberTypesettoInternational.

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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CallClassification

Callsusingthisroutepatterncanbeclassifiedasonnetoroffnetcalls.Thisroutepatterncanbeusedtoprevent tollfraudbyprohibitingoffnettooffnetcalltransfersorbytearingdownaconferencebridgewhennoonnet partiesarepresent. Whenthe"Allowdeviceoverride"boxisenabled,thecallsareclassifiedbasedonthecallclassificationsettingson theassociatedgatewayortrunk.Forexample:ifPatternyouhaveCallClassificationONNETandGatewayyou haveCallClassificationOFFNET,resultofthatcallwillbeclassifiedas:OFFNETifallowdeviceoverrideischecked.

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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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 COR List after-hours can be used to prevent toll fraud

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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Amediaresourceisasoftwarebasedorhardwarebasedentitythatperformsmediaprocessingfunctionsonthe datastreamstowhichitisconnected.Mediaprocessingfunctionsincludemixingmultiplestreamstocreateone 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, terminationofavoicestreamfromaTDMcircuit(coding/decoding),packetizationofastream,streamingaudio (annunciation),andsoforth

Different Types of Media Resources

Media Termination Point
MTP
x1000

Music On Hold
x6000

Conferencing

H.323v1

V
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

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

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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:

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Conference Controller presses Meet-Me button; gets dial-tone and dials conf call number; all conf call attendees calls conference call number.

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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 Can be used to add supplementary services

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MTPs connect to media streams using the same codec:

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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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DSP Farm Configuration Example

TOR
Cisco Unified Communication s Manager 135.1.100.20

Montreal IP WAN

Router1

Router2

sccp ccm 135.Y.100.20 sccp local FastEthernet 0/0 sccp Phone1-1 Phone1-2 3001 3002 voice-card 0 dsp services dspfarm

sccp ccm group 22 associate ccm 1 priority 1 associate profile 5 register CFGVCBCONF

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Phone2-1 5001 PSTN dspfarm profile 5 conference codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 1 associate application SCCP

Phone2-2 5002

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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

X
PSTN
CallManager cluster

A B
SFO

IP WAN
Conf

Tor Site

Caller X calls ANo voice across WAN A conferences B in 3 media/voice streams across WAN No conferencing during WAN failures

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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
1. SFO 2. TOR
CallManager Cluster

MRGL
1. TOR

PSTN

A B
Device Pool
Conf

IP WAN
Device Pool

Conf

SFO

MRG=SFO MRG=TOR

TOR

Conference between A, B and X No voice across WAN Requires extra hardware at branch No conferencing during WAN failures

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MRG = Media Resource Group MRGL = Media Resource Group List

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By deploying a local Conference Bridge for Branch office, all media stream will be local when there is a conference resource. Callmanager. First configure the local router as a Conference Bridge and then added to 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

Enable Multicasting. Without this, Multicast will not work Regardless if the router and CCM severs are configured or not
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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 Media Resource Group

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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.)

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Holder decide which music file holdee will listen but holidee decide which server it will receive the stream from.

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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.

MoH multicast server configuration

Uploading New Music File

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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

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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.

MoH multicast server configuration

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

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Ensure that MAX HOPE is set correctly.

It is usually over 2.

MoH multicast server configuration

Add the multicast MoH server in a MRG

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Multicast can only be enabled if it is selected in group level.

Enable multicast on the group otherwise multicast will not Work.

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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

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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.

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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

Similar to Route Lists and Route Groups

Media Resource Manager

User Needs Media Resource

Assigned to Device
Media Resource Group List

1st Choice Media Resource Group 1st Choice 2nd Choice

2nd Choice Media Resource Group 1st Choice 2nd Choice

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Media Resource 1 Media Resource 2 Media Resource 3 Media Resource 1

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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

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.

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Local MoH Source

Local Cisco Router can be used as a Music source Require SRST configurations 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
Loopback address of the router

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X depends on what Codec you are using G.711 is 1, G.729 is 3 etc.
addressandportnumberconfiguredforCiscoCallManagermulticastMOH.

Source Subnet of the IP Phone

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CiscoSRSTgatewayscanbeconfiguredtomulticastRealTimeTransportProtocol(RTP)packetsfrom flashmemoryduringfallbackandnormalCiscoCallManageroperation.CiscoCallManagermustbe configuredformulticastMOHinsuchawaythattheaudiopacketsdonotcrosstheWAN.Audio packetsarebroadcastfromtheflashmemoryofCiscoSRSTgatewaystothesamemulticastMOHIP

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

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NOTE: Cisco SRST multicast MOH supports G.711 only

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High Availability (SRST & AAR)

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Voice Lab Sample Topology

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deployed. SRST provide basic Phone functionality.

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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

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Survivable Remote Site Telephony (SRST)

Normal Operation
Data Backup Data
CallManager Cluster

Signaling Traffic

Traffic Signaling Traffic

IP WAN

SRST Router SFO Site

Voice Traffic
PSTN

Central Site

Voice Traffic 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.

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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 Mandatory commands which define the max-ephone 24 max. # of IP phones and directory max-conferences 8 numbers (DNs) supported by SRST. time-format 24 Default is 0 limit-dn 7960 2 ` dialplan-pattern 1 14086X6... extension-length 4

Limit maximum # of DNs assignable to particular types of phone.

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Global prefix which maps full e.164 called number to local ip phone extensions. In this case, if the DID of an inbound call is 14086X65001, it 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.

strict-match option enables strict ip address verification of IP phones trying to register to SRST router.

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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 dont.

SRST MGCP Fallback to H.323

CallManager Cluster

PSTN
FXO

Gateway
FXS

PRI

IP WAN MGCP Signaling + PRI Backhaul

TOR

SFO

MGCP

H.323

Under normal operation, the gateway translates FXS/FXO signaling into MGCP and backhauls L3 PRI signaling to CallManager When the WAN fails, the gateway reverts to H.323 operation SRST provides backup for the IP phones
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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 global service alternate DEFAULT dial-peer voice 1 pots incoming called-number . direct-inward-dial port 1/0:23 ! ! 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 Allows MGCP gateway to fall back to H.323 mode Enables gateway to fall back to default call application (H.323) when mgcpapp is not available

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Pots dial-peer for outbound calls in SRST mode. Note that it must have direct-inward-dial, otherwise, inbound PRI calls will get a secondary dial-tone

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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.

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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

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Cisco Unified SRST can send and receive voice-mail messages from Cisco Unity and other voice-mail systems

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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Example of Alias command

Alias 1 5 to 50
Any calls to 5XXX That are not registered Will be routed to 5001

alias 1 5 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 max-dn 4 preference 2

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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 lets 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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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

Call #1 Call #2

IP WAN
Call #3

X X X

CallManager

Call #3 Causes poor quality for ALL calls Many tools to give voice priority over data. Call admission control is about preventing voice oversubscription.

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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.

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Bandwidth used for CAC Bandwidthused for CAC

Codec type CODEC Type CallManager Locations Gatekeeper

G.711 64Kbps 80Kbps 128Kbps

G.729 8Kbps 24Kbps 16Kbps

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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 router

PSTN
CallManager Cluster

IP WAN
TOR Location: None Bandwidth: Infinite

SFO

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UK

Location: SFO Bandwidth: 96

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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.

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Location Configuration

Location is then assigned to devices or Device Pool

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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.

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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

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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 devices AAR Group value, and terminating DN or devices 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

Under Line Level, assign the AAR Group

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Each IP Phone or device must have AAR CSS defined on device level

How does AAR work? (cont.)

AAR Instructions
The call from Phone 1 in TOR to phone A in SFO is blocked due to Insufficient Bandwidth between CallManager Locations CCM retrieves the External/Alternate number for the terminating DN (derived from external phone number mask, e.g. 4086Y6XXXX) CCM prepends the AAR Group prefix (from Tor to SFO, e.g. 91)

Called Number
5001

4086Y65001

914086Y65001 914086Y65001
PT-TOR-LD

CCM reroute the call using AAR CSS via gateway and PSTN (assuming there is DID on the terminating side)

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9.1[2-9]XX[2-9]XXXXXX

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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

CCIE VOICE diagram Information Sheet containing DN, IP Address etc

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Cisco Unified CM Voice-Mail Integration

Cisco Unified CM Cluster PSTN

Cisco Unity Connection

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:
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 193 the call appropriately.

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CiscoUnityConnectionsupportsmessagingredundancyandloadbalancinginanactiveactiveredundancymodel consistingoftwoservers,aprimaryandasecondary,configuredasanactive/activeredundantpairofservers, whereboththeprimaryandsecondaryserversactivelyacceptcallsaswellasHTTPandIMAPrequests. BothCisco UnityandCiscoUnityConnectionSIPtrunkimplementationrequirescallforkingformessagingredundancy functionality.

Voice-Mail Integration Parameters

Cisco Unified CM PSTN Cisco Unity Connection

Cisco Unified CM parameter Number of Voice-Mail Ports Message Waiting Information Voice-Mail Port Name Line Directory Number Hunt List, Hunt Pilot, Voice-Mail Pilot, Voicemail Profile

Cisco Unity Connection parameter Number of Voice-Mail Ports MWI on/off Extension CallManager Device Name Prefix Subscriber Extension -

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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 SCCP Incoming call Cisco Unity Connection Directory, Prompts, Messages

MWI on Extension MWI off Extension Voice-Mail Pilot Hunt Pilot CM Hunt List
Voice-Mail Port1 Voice-Mail Port2 Voice-Mail Port3 Voice-Mail Port4 SCCP SCCP SCCP SCCP

forwarded call

MWI on call

Cisco UnityC Message access Directory access Prompt access Voice-Mail Port 1 Voice-Mail Port 2 Voice-Mail Port 3

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Signalling Traffic

Voice-Mail Port 4
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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

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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

Select Voicemail Profile

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Define Pilot Number and associate with a Voicemail Profile. Multiple Pilot # can be configured.

Step 2 MWI

Select Message Waiting

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Step 3 VoiceMail Port

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Number of voicemail port will depend on license.

Step 3 VoiceMail Port (contd)

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Step 3 VoiceMail Port (contd)

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Run the Voicemail port wizard. Ensure that all the settings are define as per the requirement such as

Step 4 Voicemail Hunt List

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Voicemail Hunt list should include the Line group created by the voicemail port wizard application.

Step 5 Hunt Pilot

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Hunt Pilot number and the voicemail pilot number should be the same in most cases.

Step 5 Call Forward Setting

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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

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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

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In unity connection defines the Phone System.

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Define UCM and AXL Users

Define the IP address of CallManager and port number is 143 Unity Connection use IMAP port Username and password must be the necessary privilege

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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

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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

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You can verify the unity integration by listing the voicemail port from CallManager. Port status should be registered.

Voicemail Subscriber

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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

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

**Both CLI and GUI Allowed in Lab

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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

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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 ! dial-peer voice 6000 voip destination-pattern 66.. session protocol sipv2 session target ipv4:135.X.67.230 codec g711ulaw no vad
!
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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 (contd)

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 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

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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

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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

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Here you will define the IP address of the CME router and the username and password.

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Unity Express Wizard Step 3

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Under Call Handling you must define the Voicemail Pilot Number and MWI Numbers

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Unity Express Wizard Step 4

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Select the extension and users whos mailbox you want to create. not select Set CFNA/CFB.

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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

Select the user ID that you wish to create a mailbox for. You can make these users an administrator of Unity Express as well.

Unity Express Wizard Step 5

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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

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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

Unity Express Wizard Step 7

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Save the information and logout.

Unity Express : Phone

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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

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Use this dialog box to configure the phone parameters.

Unity Express User Configuration

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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

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Define mailbox size, caller message size in seconds etc.

Cisco Unified Contact Center Express

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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)

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Agent and Supervisor functions Agent Monitoring and recording
Maybe add better descriptions here and split across twos ???

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UCCX Call Flow

JTAPI Provider = CCM IP address RmCM Provider = CCM IP Address Trigger is registered with CCM as a CTI Route Point

Normal user JTAPI user CCM CTI Route Point 3500

UCCX

CTI Port Known as Call Control Group DN 3201 - 3203

3500 Trigger Cisco Media Application

RmCM User

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Script Queue CSQ

3001 A1

5001 A2

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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.

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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

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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 dont 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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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

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 dont.

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

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 various desktop clients and applications

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Presence 7.0 Overview (contd)

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 Accept Accept Accept Presence Subscription Out-of_Dialog REFER Unsolicited Notification 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 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 doesnt 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)

Step 7 CallManager: 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.

Step 8 CallManager: 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

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 companys 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 phones lines this is the only line CUPC can control Create the end user account, and designate the users 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.

Call Manager Configuration for CUPC Add CUPC Device (contdt)

Step 12

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 users ID in the form; UPC[0-9A-Z]{1,12} meaning that you take out anything that nots a letter or number and only use the first 12 characters and capitalize them Example: user id fkhan becomes; UPCFKHAN 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 the hostname of UCM And IP address of CallManger 7.0

Define username And password of AXL User Ie. Administrator

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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

Security Key defined During installation

Define username And password of AXL User Ie. Administrator

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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 take effect. To restart the proxy server, select Presence > Routing > Settings

You must restart the SIP proxy server before any changes that you make to the application listeners 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

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 Name Profile Name 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 Protocol Type TCP * LDAP Server should be V3 compatible & anonymous read access is sufficient LDAP Server Configuration if Global Catalogue server is used for LDAP. Port 3268 Protocol Type TCP 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 Name Profile Name 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 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

Enabling QoS in the Campus

Pream. SFD DA SA Type 4 TAG Bytes PT Data FCS

hree Bits Used for CoS 802.1p User Priority)

Ethernet Frame

PRI

CFI

VLAN ID CoS

802.1Q/p Header

Application
Reserved Reserved Voice Bearer Video Conferencing* Call Signaling

802.1p user priority field also called Class of Service (CoS) Different types of traffic are assigned different CoS values CoS six and seven are reserved for network use

7 6 5 4 3

* Including Audio and Video

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High Priority Data Medium Priority Data 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

1
Switch Detect IP Phone Switch trust IP phone

PC VLAN = 10

Phone VLAN = 110

TRUST BOUNDARY

CoS 5 = DSCP 46 Voice = 5, Signaling = 3 CoS 3 = DSCP 24 CoS 0 = DSCP 0 All PC Traffic Is Reset to CoS 0 PC Sets CoS to 5 for All Traffic

2 3

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Typical Cisco IP Phone will generate traffic with the following tag o o o Signaling CS3 or DSCP 24 Media/RTP EF or DSCP 46 PC Traffic will be over written by the IP Phone to 0

Classification and Marking Design

QoS Baseline Marking Recommendations

Application Routing Voice Video Conferencing Streaming Video Mission-Critical Data Call Signaling Transactional Data Network Management Bulk Data Scavenger L3 Classification IPP 6 5 4 4 3 3 2 2 1 1 0 PHB CS6 EF AF41 CS4 AF31* CS3* AF21 CS2 AF11 CS1 0 DSCP 48 46 34 32 26 24 18 16 10 8 0 L2 CoS 6 5 4 4 3 3 2 2 1 1 0
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Best Effort

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 CallSignaling 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 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)#

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.

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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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List of Port to remember in order to create access list based on certain traffic type.

WAN Edge QoS Design Considerations

Campus Distribution/Core Switches

Queuing/Dropping/ Shaping/Link-Efficiency Policies for Campus-to-Branch Traffic

WAN Aggregator

WAN
LAN Edges WAN Edges

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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 video5 to 11 Class (QoS baseline) model

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CRTP,orRTPheadercompression,isamethodfordecreasingthesizeoftheVoiceoverIP(VoIP) packetheaderstoreducethebandwidthconsumed

CRTPwasdesignedforreliablepointtopointlinkswithshortdelays

WAN Edge Bandwidth Allocation Models

Three-Class (VoIP and Data Only) WAN Edge Model

Best Effort (62%)

Voice 33%

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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 bandwidth percent 5 ! Minimal BW guarantee for Call-Signaling class class-default fair-queue ! All other data gets fair-queuing !

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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 Speed
56 kbps 64 kbps 128 kbps 256 kbps 384 kbps 512 kbps 768 kbps Fragment Size

CIR

Bc

53500 bps 60800 bps 121600 bps 243500 bps 364800 bps 486400 bps 729600 bps

532 bits per Tc 608 bits per Tc 1216 bits per Tc 2432 bits per Tc 3648 bits per Tc 4864 bits per Tc 7296 bits per Tc

70 Bytes 80 Bytes 160 Bytes 320 Bytes 480 Bytes 640 Bytes 960 Bytes

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Frame Relay QoS Design

WAG FR Link 768 kbps Frame Relay Cloud BR

<a 3 to 5 Class Model can be used> Optional: Enabling Class-Based cRTP

! 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 ! 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 !

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In order to enable FRF 12 ensure frame relay class map has frame-relay fragment command with the right value

MLPoFR QoS Design

WAG <a 3 to 5 Class Model can be used> Optional: Enabling Class-Based cRTP ! interface Serial6/0 no ip address encapsulation frame-relay 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 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 ! MLPoFR Link 768 kbps Frame Relay Cloud BR

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Unified Mobility

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Cisco CallManager Extension Mobility Overview

Log on to a Cisco IP Phone 7940 or 7960 in a Cisco CallManager cluster to get extension Device profile includes: extension, services, class of service restrictions applied to IP Phone Login modes: Auto-logout other IP Phones Keep login on other IP Phones Logout modes: Explicit logout at IP Phone Timed logout

Single Cluste r
IP LAN

User Office IP Phone 7960 (x5000) LDAP Directo ry

IP Phone Service s CRA Server

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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

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Under Service parameter you must define some EM parameters such as allow multiple login

Device Profile Default Configuration for the Cisco IP Phone 7960

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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

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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

Unified Mobility Single Number Reach

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Cisco Unified Mobility

Cisco Unified Communication s Manager Customer

Call to office number.

Call to Mobile Voice Access directory number.

PSTN
Gateway Remote Phone Office Phone

Mobile Voice Access establishes a system to create enterprise calls from any location. Mobile Connect lets remote and office phones ring simultaneously.

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With Mobile Connect, calls placed to office phones ring the office phones and associated remote phone.

Cisco Unified Mobility has two components: Mobile Connect and Mobile Voice Access (MVA).

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

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Call logging (CDR)

Cisco Unified Communications Manager user webpages

Access to enterprise features from remote phones using DTMF: Softkeys can be used on phones with smart client installed.

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Mobile Connect Call FlowIncoming Calls to Office Phone

Cisco Unified Communication s Manager

Gateway 5145553XXX 4165551555

Outside Caller

Mobile Connect

Call to 1-514-5553001

PSTN
Caller ID: 416-555-1555 Office Phone 3001 604-5552002

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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.
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

Remote Phone of 3001

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Cisco Unified Mobility Configuration Elements

Configuration Element Name
End User

Configuration Element Function

The end user is referenced by the office phone and remote destination profile. Mobile Connect and/or MVA must be enabled. A maximum number of remote destinations can be configured. The office phone needs to be configured with an owner (i.e., the end user). A virtual phone device. Per office phone number, a shared line is configured. End user, (device) CSSs, and MOH audio sources are specified. One or more remote destinations are added. Associated with shared line(s) of remote destination profile. Configured with destination number. Optionally, access lists can be applied. Mobile Phone and Mobile Connect functions are selectively enabled. Filters used to permit or deny incoming calls placed to the office phone to ring a remote destination. Permitted or denied caller IDs are specified. Media resource used to interact with the VoiceXML call application running on a Cisco IOS router. Only required for MVA.
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Phone Remote Destination Profile Remote Destination

Access List MVA Media Resource

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Shared Line Between Phone and Remote Destination Profile

Line1: 3001 Partition CSS etc. Office Phone 1 MAC Address Owner CSS etc. shared line Call to shared line rings office phone line and remote destination(s ) associated with corresponding line of remote destination profile.

Line1: 3002 Partition CSS etc.

Office Phone 2 MAC Address Owner CSS etc. shared line

Remote Destination1 : 914168391717 Remote Destination2: 9011971380523 0

Line1: 3001 Partition CSS etc. Line2: 3002 Partition CSS etc.

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Remote Destination Profile User ID CSS Rerouting CSS etc.

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

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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

One of the destination where phone will ring when Some one is calling the user at their desk phone

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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 users 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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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

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Exam Tips
Read the entire exam Redraw your topology Time management Clarification Make notes Check list Unexpected items Troubleshooting (10-minute rule) Functionality testing

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Test-Taking Strategies
Arrive early or visit the site the day before Dont schedule flights too close to the end of the examit can run overtime Get some sleep the night before the exam

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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

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Test-Taking Strategies (Cont.)

Do each question as a unitconfigure and verify before moving to the next question. But do not spend more then 5 minute in troubleshooting Dont assume requirements that arent mentioned in a question Dont make any drastic changes in the last half hour of the exam Save your configs often

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Test-Taking Strategies (Cont.)

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 device and work on something else while it comes back up in a known state.

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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-- Register Phones - 15 mins Media - 15 mins

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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)-- Testing - Rest

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