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RouteHub Group, LLC Page 1 www.routehub.net

Deploying Cisco UC

Manager Express andUnity ExpressVoice & Unified Communications: Small Business

Practical Cisco Training for Network Engineers & Consultants!

RouteHub Group, LLCwww.RouteHub.net

June 30, 2010
http://www.routehub.net/http://www.routehub.net/http://www.routehub.net/


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Table of Contents

1 Introduction 82 Concepts 93 Design 11

3.1 Our Design Small Business Voice Design 113.2 Requirements 133.3 Solutions and Topology 133.4 Topology Services and Sub-Services 143.5 Hardware & Software 153.6 Network Diagram 16

4 Configuration for CME 174.1 Initial Configuration 174.2 CallManager Express 18

4.2.1 Telephony Service (telephony-service) 184.2.2 Directory Number (DN) Configuration (ephone-dn) 234.2.3 IP Phone Configuration (ephone) 24

4.3 Voice and Data VLAN Configuration 254.4 Configuring DHCP on Cisco IOS 274.5 Mapping an analog line (DID) to an IP phone 274.6

Configuring FXS port as a SCCP port 30

4.7 CME as SIP Server for SIP Clients 324.8 Blocking incoming calls from PSTN 334.9 Setting up an Authenticated SIP Trunk to SIP Provider 354.10 Phone Directory 384.11 Single Number Reach (SNR) 384.12 Sending Calls to Voicemail (CUE) 394.13 Conferencing 41

4.13.1 MeetMe Conferencing 414.13.2 Ad-Hoc Conferencing 44

4.14 Paging 454.15 Personal Speed Dial 464.16 Upgrading CallManager Express 464.17 Intercom 474.18 Hunt Group 484.19 Call Park 484.20 How to setup Phone Softkey templates 494.21 Extension Mobility 50


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4.22 How to setup a custom ring tone? 524.23 Call Center 554.24 Fax to Email using T.37 584.25 Phone Services 614.26 Cisco CME using Exchange 2007 UM 624.27

Using a XML Menu File for Phone Services 64

4.28 Installing SIP Firmware on Cisco 7940/7960 664.29 VoiceView 714.30 Installing SIP Firmware on Cisco 7941/7961/7962+ 804.31 Cisco Unified CallConnector 84

4.31.1 Server Installation 854.31.2 Components 1094.31.3 Adding a new user 1114.31.4 Client Installation 1174.31.5 Using Cisco Unified CallConnector 124

5 Configuration for CUE 1345.1 Access to CUE 134

5.1.1 CME Configuration 1345.1.2 Console into the CUE Service Engine. 136

5.2 Unity Express 1365.2.1 CUE Global Configuration 1375.2.2 Enable Voicemail Services 1385.2.3 Sending Calls to Voicemail on CME 1395.2.4 Create User Voice Mailboxes 1405.2.5 Enable other CUE services (like Auto Attendant) 141

5.3 Upgrade CUE to Version 7.x 1425.4 Coping Files to CUE via CLI 1455.5 Auto Attendant 1465.6 Voicemail Email Notifications 1495.7 CUE and CME on separate routers 151

5.7.1 CME Router 1515.7.2 CUE Router (Cisco CUE Router Configuration) 1545.7.3 CUE Router (CUE Configuration) 155

5.8 Live Record 1585.9 Downgrade CUE software 1605.10 Basic CUE Start-Up Wizard 167

6 Monitor 1706.1 Operations 170

6.1.1 IP Phones 1706.1.2 Conferencing and DSP resources 1726.1.3 Dial Plan and Cisco CallManager Express 177


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6.1.4 SIP 1826.1.5 External Calling summary 1916.1.6 Email Notification and Voice Messaging (CUE) 192

6.2 Troubleshooting 1946.2.1 Root Causes 1946.2.2

Initial questions to ask 194

6.2.3 Typical fixes 195

7 Sample Full Configuration 1967.1 CME and CUE on UC520 196

7.1.1 CME 7.1 on UC520 1967.1.2 CUE 7.0.1 on UC520 216


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1 Introduction

Many site focused on providing training towards certifications or exams. These are importantfor career development and we have CCIE, CCNP, and CCNA certifications. So we know

that they are very valuable to your network engineering career, however, they do not teach

practical network training relevant for network engineers and consultants in the real world.

This is what our training format is based upon providing practical solutions and technologiesthat are deployed in real working environment. Our training workbooks provide the fourmajor components:

Concepts

Design

Configuration

Monitor

Learn the concepts that matter in terms of the components and protocols involved for atechnology's operation.

Learn how to design a network solution with practical steps, considerations, and tools foryour company or clients.

Learn how to configure a network with best practices and get operational step-by-step. Wealso include full working configuration files for our workbooks.

Learn how to monitor, troubleshooting, and confirm the operational state of your configurednetwork.

All four are important for network engineers and consultants to know how to manage anetwork in real time.


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2 Concepts

A Cisco IP Telephony solution really has two categories when discussing which option is bestfor a customer; Enterprise and Small Business. Enterprise Cisco IP Telephony would involvethe Cisco Communications Manager (previously known as Cisco CallManager) and CiscoUnity for Unified Messaging. Small Business Cisco IP Telephony would involve CallManagerExpress and Cisco Unity Express, taking full advantage of their existing Cisco router for othernetwork services (e.g. Internet connectivity, Firewall, and Remote Access capabilities suchas SSL VPN and Client-based IPSec VPN access).

Why would a Small Business use a Cisco solution? Well, many business owners arecautious to place all their eggs in one basket in general, but companies know that Cisco is asolid company with a strong networking focus to their businesses model. Now, manyproducts that Cisco has offered to their customers were really designed for medium to largebusiness due to capabilities with performance, security, and reliability aspects that are

important to a business. Well because of those capabilities a high price tag is associatedwith it. Then again, you get what you pay for. I have been deploying Cisco networks formore than 12 years and with Cisco hardware deployment (design and implemented correctlyof course), it just works! I rarely touch or revisit a Cisco product implementation for continuedsupport or troubleshooting. Hence, why many companies tend to choose Cisco for thisreason among other critical requirements needed in a solution.

Cisco's Enterprise IPT solution is tailored for environments, which require 2500 to 10,000 IPendpoints. This would involve many Communication Manager Servers deployed in a clustersetup, which is very clean and provides ease of administration. Tracing call activity with SDLfiles for example can be a little tricky at times. For small business, going with theCommunication Manager product is an overkill, not needed, and very expensive. SmallBusiness tends to turn to NEC, for example, for a Small Business IP Telephony solution.

Another alternative with the Cisco Small Business product line is CallManager Express andCisco Unity Express. They provide scaled down capabilities of the Cisco Communication'sManager product, but offer its full capabilities for call routing and voicemail for customers in asingle solution, not turnkey style!

Basically, as a consultant, the Cisco router must be setup and configured for CallManagerand Unity initially. Administrating IP Phones, extensions, etc can be done via the web portalprovided for the IP Telephony Express Suite (CallManager and Unity Express). I have donedeployments where we have a Cisco 2800 series router with IOS 12.4 running with CME(CallManager Express) capabilities and a AIM-CUE (providing Cisco Unity Express) installedplus a Wireless 802.11g WIC card to provide Wireless capabilities. With 12.4 we can setupWebVPN or SSL VPN for users. We can setup DMVPN for hub-and-spoke VPN capabilitiesto different sites and not compromising security. We can enable a stateful firewall on the

router that would connect to a DSL or Cable Modem device for Internet Access. To evenproviding DHCP services for local LANs to Quality of Service (QoS) to preserve voice quality.This is one design option for deploying Cisco CallManager Express and Unity Express to aSmall Business with those added capabilities. It becomes more cost efficient and it's robust.So, that is the design (one way at least) for how you can use this solution.

However two issues arise, 1) what about Small and SMB sizes, and 2) Turn-key solution!For organizations between 250 to 500 users, CallManager and Unity Express is not anyoption any more. Therefore, a Cisco Communication Manager is required, hence high costand endless features going unused. Cisco in the past year started developing products and


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solutions tailored for Small and SMB size customers. There is now a Cisco CommunicationManager appliance designed for Small Businesses in a single turnkey solution providing callmanager, voicemail, and other great functionality. This part of Cisco CommunicationManager 6 product line is the "Business Edition" and is designed to support up to 500 usersand endpoints. Another SMB turnkey solution is the Unified Communications 500 Series.Remember what we discussed in terms of using a Cisco 2800 series for example andintegrating CME, CUE (via NM-CUE or AIM-CUE), Wireless, etc.

Well, I think Cisco has been reading all of our minds. The Unified Communications 500device provides all of these options in one single turnkey solution. The UC500 can beconfigured via CLI or through a new GUI application for easy administration. The choice isup to you as a consultant or engineer. I have deployed numerous UC520 products to manySOHO and Small size businesses providing a robust voice solution. Plus it's a lot quieterthan having a 2800 or higher running in your facility.

So, when it comes to designing an IPT solution choosing the right solution is based on the

size of the environment, growth considerations, and functionality required by the customer.

SOHO to Small networks would normally get CallManager Express, Unity Express, or the

Unified Communication 500 device. Small to SMB networks would be border line with

CallManager Express & Unity Express, but Cisco Communications Manager 6 BusinessEdition would be a better fit especially when potential growth comes into play. And for

Medium to Large Enterprise, the full blown version of Cisco Communication Manager

deployed in clusters would be recommended.

In terms of the Cisco Communication Manager family series, there is CallManager 4,

Communication Manager 5 (previously called CallManager 5), and now Communication

Manager 6. Communication Manager 5 and 6 are pretty much the same except for a few

enhancements provided in the 6 release such as integrated Unified Presence within the

product than separating it out plus providing great Mobile capabilities allowing your cell

phone and office phone to ring at the same time.


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3 Design

3.1 Our Design Small Business Voice Design

Our network design will include Voice and Unified Communication user services. Below arethe details on how our voice network is designed.

RequirementsOur network consist of 4 users with a potential growth up to 8 users (a Small Business) eachwith an IP Phone, but our configuration will only show a handful of our phones plugged in.Our voice network will use a single analog line from their local PSTN provider with adedicated phone number (aka DID number). The small office will require use of all basicvoice features such as conferencing, redial, speaker phone, etc.

Voicemail is also needed for our users and ideal if there voicemail messages can be sent viaemail. Other unique features with the voice solution would also be great, but not required.They only have a single location and will require some form of remote access to access theoffice's resources remotely.

Some of the requirements and expectations include:

Voice system with voicemail application user services for the office

Remote access to resources

Solutions and TopologyBased on our requirements, our applicable solutions for our environment are the following:

LAN; required because each site has a LAN network where all servers, desktops,and IP phones would connect into for access to other user services.

Our LAN topology, our hierarchical design, would be a single tier giving us a LAN CollapsedCore model since our office consists of less than 24 devices.

Our general design with our solution will consist of a single LAN subnet using the followingschema: 10.67.78.Y /24 configured on our LAN Collapsed Core; where "Y" is designated forthe node.

Bandwidth services within our LAN will be FastEthernet.

Topology Services

Within our LAN topology we will utilize the following network services applicable to ourenvironment and requirements:

Required Services

Routing & Switching: Static routing; routing is required and since we do not havemultiple sites or routing devices we would use default route with the ISP.

Security (VPN): SSL VPN; provides remote access services for users to access theoffice resources remotely using HTTPS/SSL VPN.


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As for our user services, we will implement Voice and Unified Communications on our LANnetwork with a call processing and voicemail solution to support up to 8 users. Using voicerequires our LAN for additional network services such as a Multicast, Quality of Service(QoS), VLANs, and 802.1q Trunking.

HardwareWe need hardware to support our topology services, user services, bandwidth services, andrequirements in our design that includes the following:

Voice: Call Processing & Voicemail support up to 8 users with basic voice features

Multicast

Quality of Service (QoS)

Static Routing

SSL VPN

VLANs

802.1q Trunking

FXO port for the analog line

In our network we can consolidate our LAN Collapsed Core and our voice user servicestogether for simplicity.

The hardware chosen for our design will consist of the following:

Hardware: Cisco UC520 (license for 8 users) integrated with Voicemail serviceengine

o Cisco UC Manager Express (using OS 7.x)o Cisco Unity Express (using OS 7.x)

Software: IOS 12.4 Advanced IP Services Feature set

Cisco IP Communicator

Cisco IP 7970 phone

Third-Party SIP phone

Our Cisco UC520 appliance allows us to send voicemail messages via email to access withother voice features to include some of the following:

Extension Mobility

Single Number Reach (SNR)

Live Record

Paging, Intercom

SIP Services

and more!


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3.2 Requirements

First, we need to determine all the business and technical requirements. Understand what isneeded, the expectations involved, budgetary considerations, network services, securityregulations, and more much outlined by the company or business

We would gather details for building our design based on the following: Requirements and Expectations

Traffic

Budgetary Considerations

Existing Components and Services

Technical Objectives

The technical objectives are what define best practices and recommendations in a networkdesign. These are often challenges that many networks face early or further down the roadwith a network. When there are issues its usually due to one of the objectives that were nomet or considered during the design phase.

Below are the technical objectives our design should consider, include, and bring up with therequirements gathering:

Performance

Reliability

Scalability

Security

Flexibility

Network Management

3.3 Solutions and Topology

Once the requirements and objectives have been gathered, that info will help with the designprocess of our solutions and topology.

At a high level the solutions is the network that deals with a specific function or task based onthe requirements gathered. Many network solutions listed here do require the existing ofother solutions to work. The one network solution that is required for all solutions is the LANsolution which is essentially the network backbone that connects all the other solutions

together.

Below are the solutions we can choose from.

Local Area Network (LAN), Wireless

Wide Area Network (WAN), Metropolitan Area Network (MAN)

Internet Edge

Data Center


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Once the solutions have been determined it is time to build our topology. The topology isbasically the framework in our design that doesnt contain any technologies, services,protocols, or hardware devices by name yet. We are essentially just building a street withnothing on it.

There are many ways to build a design and usually common topologies and case studies areoften used.

These topologies really include tier levels in the design. One way to explain is with a LANtopology which is often discussed in many networking textbooks. A best practice andrecommended LAN would consist of a LAN Core, LAN Distribution, and LAN Access. This isa tier level model consisting of 3 tier levels, each with a certain ideal purpose.

A LAN Access provides direct access to nodes like computers, printers, IP Phones, accesspoints, etc. LAN Distribution deals with aggregating the traffic from the Access layerincluding other roles with routing, switching, and security policies. And the LAN Core is seenat the backbone where the LAN Distribution connects into providing high-speed switchingand forwarding. This three tier model accommodates much of the technical objectivesespecially with scalability and reliability among others. But a 3-tier model is often seen withlarger networks.

Some solutions typically can have 1 or 2 tiers in most designs. Again 3 tier designs are oftenseen with large size networks or very large networks. But some of the tier levels can beconsolidated where needed and the hardware that you choose that can also change the tierlevel in the design. For example, an Internet Edge solution typically consists of 3 tiers (theEdge Router, the Edge Switch, and the Perimeter Firewall). Well nowadays the edge switchhas been eliminated being integrated with the Edge Router leaving us with a 2 tier model,which is the most common, however, the firewall services can also be integrated with ourEdge router that provide stateful firewall inspection with capabilities such as rACL (ReflexiveACL) or CBAC. Thus, our Internet Edge device can be a 1 tier model.

2 tier models are very common for small and medium sized networks.

3.4 Topology Services and Sub-Services

Once the topology has been determined (or narrowed down), the next thing to determine isthe topology services that will overlay on-top of our topology.

This can include the following services:

Routing & Switching

Security & VPN

Tunneling

Voice & Unified Communications

Wireless

Other Technologies (like QoS and HSRP)

Topology sub-services deals with the extended features within the services within thenetwork design.

For example, one of our topology services could be Routing using OSPF. Well OSPF hasmany design considerations and best practices that can include configuring routesummarization within a LAN Distribution to send summary routes up to a LAN Core. A


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common best practice discussed with OSPF including Stub routing within the LAN Accessnetwork among other sub-services.

For example, MPLS, which is a topology service, these are sub-services that can bedeployed with MPLS.

General

Route Reflectors

VRF Selection Traffic Engineering (TE)

Extranet

MPLS over GRE, MPLS over DMVPN

QoS service to MPLS VPN

IPv6

Internet Access service

Multicast service to MPLS VPN

3.5 Hardware & SoftwareDetermine the best hardware and software solutions for each component in the design toaccommodate the following points:

Requirements

Topology Service and Sub-Services

Business Size considerations

The hardware device can be any vendor besides Cisco. Make sure the hardware chosensupports the requirements and services in our design including considerations for thebusiness size of the network and the technical objectives.


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3.6 Network Diagram

Below is the network diagram showing our completed design with voice user services.


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4 Configuration for CME

4.1 Initial Configuration

The first we need to do is console or connect into each device on our network based on theinformation presented in the network diagram.

Second, complete all basic configurations for all devices based on the following:Configure all interfaces based on the network diagram in terms of IP addressing and thesubnet mask.

Next enable all interfaces by issuing a no shutdown

Once that has been completed we need to check on two things.

First confirm that all interfaces are up and running. This command will show all interfacesand there status in a basic or brief view. Confirm that all interfaces once configured showsan UP UP status.

show ip interface brief

And second, confirm basic network connectivity by pingingthe directed connected IP addressof the other router. Do this for each device.


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4.2 CallManager Express

Cisco CallManager Express or Cisco Unified Communication Manager (UCM) Express is acall processing solution aimed for Small/SMB businesses. They provide many voice featuresand applications such as Call Center. Call Processing is the central component in a VoIPnetwork infrastructure where everything connects such as IP Phones, Voice Gateways, and

other external voice applications like voicemail.

Before any of the features below can be configured we need to enable Cisco CallManagerExpress (CME) or Cisco Unified Communication Manager (UCM) Express on a Cisco routersupported for CME.

1. We will need to configure our CME server, which is done under telephony-service.2. Next we need to configure our Directory numbers (DN), which is a unique extension

or number used by users with IP phones . This is done under what is called ephone-dn.

3. And last we associate the configured DN to a physical IP Phone on the network.This is done under what is called ephone.

4.2.1 Telephony Service (telephony-service)

To enable a basic configuration for CME is actually very simple. All the extra features addedis what makes the configuration look very long.

STEP 1:LANINTERFACE AND IPCONFIGURATION SUMMARYOur CME configuration will be on a Cisco UC520 appliance supporting up to 8 users. OurLAN interface on the UC520 where all of our IP phones and systems are connected to isconfigured as followed:

vlan 10name ROUTEHUB-VLAN

interface FastEthernet0/1/1description IP Phone Portswitchport access vlan 10

interface Vlan10no ip addressno ip redirectsno ip unreachablesno ip proxy-arpbridge-group 10bridge-group 10 spanning-disabled

interface BVI10

ip address 10.67.78.1 255.255.255.0no ip redirectsno ip unreachablesno ip proxy-arp


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STEP 2:ENABLE TELEPHONY SERVICEWe will enable CME on our router using the IP address from our LAN and specify the SCCPport number of 2000. We need to enable telephony-service first before enabling all othercommands within this section.

telephony-service

ip source-address 10.67.78.1 port 2000

STEP 3:TIMEOUT FOR INTER-DIGITSWhen calls are placed from any IP phone registered with CME it will take 5 seconds for CMEto setup the call. This is important to configure since the default timeout value is long and isa common compliant among users who are placing calls and it takes a long time for the callto get connected.

timeouts interdigit 5

STEP 4:BANNER ON IPPHONEWe can configure a short banner that would be displayed on all IP phones near the softkeys.In our configuration our banner would read RouteHub UC520.

system message RouteHub UC520

STEP 5:AUTO-REGISTRATIONWe will use auto-registration on CME where any new phone plugged into the network willautomatically get a temporary DN from a list of DN configured on CME. In our configurationour auto-registration will be the DN from profile 19.

auto assign 19 to 19

STEP 6:VIDEO SUPPORTIf video related services with Cisco VTAdvantage are used it can be enabled globally for IPphones that support video capabilities with Cisco VTAdvantage like the Cisco 7970 and 7960phone series.

video

STEP 7:TIMEZONENext we will specify the timezone that CME will refer and use for the time for all IP phonesconnected to CME.

In our configuration we will choose 5 which is for PST.

time-zone 5

Below are the numbers for other time-zone numbers we can choose from:


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1 Dateline Standard Time -7202 Samoa Standard Time -6603 Hawaiian Standard Time -6004 Alaskan Standard/Daylight Time -5405 Pacific Standard/Daylight Time -4806 Mountain Standard/Daylight Time -420

7 US Mountain Standard Time -4208 Central Standard/Daylight Time -3609 Mexico Standard/Daylight Time -36010 Canada Central Standard Time -36011 SA Pacific Standard Time -30012 Eastern Standard/Daylight Time -30013 US Eastern Standard Time -30014 Atlantic Standard/Daylight Time -24015 SA Western Standard Time -24016 Newfoundland Standard/Daylight Time -21017 E. South America Standard/Daylight Time -18018 SA Eastern Standard Time -18019 Mid-Atlantic Standard/Daylight Time -120

20 Azores Standard/Daylight Time -6021 GMT Standard/Daylight Time +022 Greenwich Standard Time +023 W. Europe Standard/Daylight Time +6024 GTB Standard/Daylight Time +6025 Egypt Standard/Daylight Time +6026 E. Europe Standard/Daylight Time +6027 Romance Standard/Daylight Time +12028 Central Europe Standard/Daylight Time +12029 South Africa Standard Time +12030 Jerusalem Standard/Daylight Time +12031 Saudi Arabia Standard Time +18032 Russian Standard/Daylight Time +18033 Iran Standard/Daylight Time +210

34 Caucasus Standard/Daylight Time +24035 Arabian Standard Time +24036 Afghanistan Standard Time +27037 West Asia Standard Time +30038 Ekaterinburg Standard Time +30039 India Standard Time +33040 Central Asia Standard Time +36041 SE Asia Standard Time +42042 China Standard/Daylight Time +48043 Taipei Standard Time +48044 Tokyo Standard Time +54045 Cen. Australia Standard/Daylight Time +57046 AUS Central Standard Time +570

47 E. Australia Standard Time +60048 AUS Eastern Standard/Daylight Time +60049 West Pacific Standard Time +60050 Tasmania Standard/Daylight Time +60051 Central Pacific Standard Time +66052 Fiji Standard Time +72053 New Zealand Standard/Daylight Time +720


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STEP 8:VOICEMAILNext we will specify what the voicemail main number will be. This basically configures aspeed dial for users who want to check their voicemail they can simply press the voicemailbutton on their phone.

voicemail 6000

STEP 9:WEBADMINACCOUNTWe can also configure CME, our DNs, and phones directly from a GUI interface which isenabled once CME is configured. But, we need to configure a username and password toaccess the GUI page. We can configure our web admin account by doing the following:

web admin system name admin secret cisco123

To access the GUI page we would simply go to a web browser use the IP address of theCME server followed by ccme.html (for example):

http://10.67.78.1/ccme.html

That will prompt for a username and password where we would input admin / cisco123.

Once we are logged in successfully we should see the following page where we canconfigure our Phones, Extensions, and System Parameters listed under Configureamongother configuration.

STEP 10:MUSIC ON-HOLD (MOH)

MOH is also enabled by default and the following file is used for MOH, which is not requiredto be configured. However, if a different MOH file is needed it would be configured here.

moh music-on-hold.au
http://10.67.78.1/ccme.htmlhttp://10.67.78.1/ccme.htmlhttp://10.67.78.1/ccme.html


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STEP 11:OTHER CONFIGURATIONOther configuration to add for CME relating to call-forwarding and transferring calls within thevoice network are the following:

call-forward pattern .Tcall-forward system redirecting-expandedtransfer-system full-consult dsstransfer-pattern 9.Tsecondary-dialtone 9

STEP 12:DEFAULT (EXAMPLE FROM CISCO UC520)A lot of defaults will be added under telephony-services not configured by the engineer, thatmay include some of the following as an example.

max-ephones 14

max-dn 56load 7914 S00104000100load 7902 CP7902080001SCCP051117Aload 7906 SCCP11.8-0-3Sload 7911 SCCP11.8-0-3Sload 7921 CP7921G-1.0.1load 7931 SCCP31.8-1-1SR2Sload 7936 cmterm_7936.3-3-5-0load 7960-7940 P0030702T023load 7941 TERM41.7-0-3-0Sload 7941GE TERM41.7-0-3-0Sload 7961 TERM41.7-0-3-0Sload 7961GE TERM41.7-0-3-0Sload 7970 term70.defaultload 7971 TERM70.7-0-3-0Screate cnf-files version-stamp 7960 Mar 10 2009 14:54:25


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4.2.2 Directory Number (DN) Configuration (ephone-dn)

Once CME has been configured, the next part is to configure the DN or extensions that wouldbe used on the phones.

In our configuration example we will configure four DN that will later be mapped between twoIP Phones on our network.

Below we will configure a DN of 1001 (configured as number 1001). If this DN is associatedto one of the IP Phone buttons that label or display for that extension would read 1001(Office). A username can be associated to this DN (configured as name 1001). If a callertries to call DN 1001 and there is no answer within 15 seconds or the line busy (because weare on another call or the phone is not registered) the call is forwarded to 6000, which is ourDN for voicemail. Below shows that configuration:

ephone-dn 3 dual-linenumber 1001label 1001 (Office)

name 1001call-forward busy 6000call-forward noan 6000 timeout 15

We will configure two for DNs using extensions 1002 and 6700:

ephone-dn 4 dual-linering internalnumber 1002label 1002 (Family Room)name 1002call-forward busy 6000call-forward noan 6000 timeout 15

For DN 6700, we will add a description that would display the external number right aboveour lines/extensions on our IP phone:

ephone-dn 10 dual-linenumber 6700label 6700 (Main)description 9252302203name 6700call-forward busy 6000call-forward noan 6000 timeout 15

The next DN will be for extension 3001 and all call received will be forwarded to DN 4001

automatically:

ephone-dn 5number 3001call-forward all 4001


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4.2.3 IP Phone Configuration (ephone)

Once the directory numbers have been configured on our CME router next we want toconfigure our IP Phones and associate specific DNs to the phone. In this example, we willuse two Cisco 7970 IP phones.

NOTE: Make sure to configure DHCP before connecting IP Phones. You canreference section 4.3 for how to configure DHCP on a Cisco IOS device.

For Phone1, that IP Phone and its MAC address will be listed under ephone 6. Its type willautomatically be provided to us once the phone starts up, so we dont need to configure that.

For button 1 on our IP phone it will use DN profile 1, which is for 6700. For button 2 on our IP phone it will use DN profile 3, which is for 1001. For button 3 on our IP phone it will use DN profile 5, which is for 3001, but it is

configured to forward all calls to DN 4001.

ephone 6mac-address 0011.932B.8B15type 7970button 1:10 2:3 3:5

For Phone2, that IP Phone and its MAC address will be listed under ephone 2. Its type willautomatically be provided to us once the phone starts up, so we dont need to configure that.

For button 1 on our IP phone it will use DN profile 1, which is for 6700. Making this ashared line now used between both 7970 phones.

For button 2 on our IP phone it will use DN profile 4, which is for 1002.

ephone 2mac-address 001C.58F0.7619

type 7970button 1:10 2:4

NOTE: If Auto-Registration is configured new phones plugged in will use the temporarynumber hence that new phone will be added with a new ephone listed in ourconfiguration including its MAC address. We can simply just locate that ephone andre-configure the buttons with its new extension.


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4.3 Voice and Data VLAN Configuration

Two of the biggest best practices and recommended discussed with all IP TelephonySolutions are with the following points:

1) Implement End-to-End QoS giving Voice RTP traffic high priority

2) Create a separate network (VLAN) for your Voice traffic

Here is the configuration for the second common point discussed.

1. First, on your L2 network configure two VLANs, one for Data and the other for Voice.

vlan 10name RHG-VLAN-DATA

vlan 100name RHG-VLAN-VOICE

2. Next we will configure a switch port that has a PC connected into an IP Phone, whichis then plugged into a switch port reflecting our DATA and VOICE VLAN assignment.Let's assume this port is for Fa0/1.

interface FastEthernet0/1description EDGE: VLAN DATA+VOICEswitchport access vlan 10switchport mode accessswitchport voice vlan 100spanning-tree portfast

3. Let's say that this configuration is on a Access Switch connecting into a Core or

Aggregation switch configured for these two VLANs. Well this uplink port needs tobe configured for 802.1Q to carry our two VLAN tags across. We will also be specificin our configuration and only allow our two configured VLANs. Let's assume thisuplink port is Gi0/1.

interface GigabitEthernet0/1description UPLINK: LAN CORE OR AGGswitchport trunk allowed vlan 10,100switchport mode trunkswitchport nonegotiatespanning-tree portfast trunk


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4. If the routing for our Data and Voice VLANs are configured on a L3-switch (likely ourLAN Core and Aggregation L3-switch in our network) then we can configure the SVIinterface for the two VLANs making them routable. Below is that configuration:

interface Vlan10description SVI: VLAN DATAip address 10.67.79.1 255.255.255.0

no ip redirectsno ip unreachablesno ip proxy-arp

interface Vlan100description SVI: VLAN VOICEip address 10.67.78.1 255.255.255.0no ip redirectsno ip unreachablesno ip proxy-arp

Configuration Summary

vlan 10name RHG-VLAN-DATA

vlan 100name RHG-VLAN-VOICE

interface FastEthernet0/1description EDGE: VLAN DATA+VOICEswitchport access vlan 10switchport mode accessswitchport voice vlan 100spanning-tree portfast

interface GigabitEthernet0/1description UPLINK: LAN CORE OR AGGswitchport trunk allowed vlan 10,100switchport mode trunkswitchport nonegotiatespanning-tree portfast trunk

interface Vlan10description SVI: VLAN DATAip address 10.67.79.1 255.255.255.0no ip redirectsno ip unreachables

no ip proxy-arp

interface Vlan100description SVI: VLAN VOICEip address 10.67.78.1 255.255.255.0no ip redirectsno ip unreachablesno ip proxy-arp


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4.4 Configuring DHCP on Cisco IOS

To enable DHCP services on our network this can be configured on a Cisco router, L3-switch, or firewall, but its recommended to enable DHCP services on a server. But, forsmall environments we can enable DHCP on our Cisco IOS router that is also running CME.

Below we will configure a DHCP scope for the 10.67.78.0 network assigning usable IPaddresses starting with 10.67.78.30 to 10.67.78.254. We will define the default gateway andDNS servers the devices (our IP Phones) would use. We will also include option 150 withthe IP address of the CME router. This is important to specify the location of our CME routeror our connected IP Phones will not know how to register with the phone system. Option 150points to the TFTP server, which happens to be our CallManager Express router. The TFTPwill supply info to the IP Phones for which Call Processing server to register with.

ip dhcp excluded-address 10.67.78.1 10.67.78.29

ip dhcp pool ROUTEHUB-DHCP-LAN-POOLnetwork 10.67.78.0 255.255.255.0default-router 10.67.78.1

option 150 ip 10.67.78.1dns-server 206.13.28.12 64.169.140.6lease infinite

4.5 Mapping an analog line (DID) to an IP phone

This is a commonly question asked, but rare to find how it can be configured with CME. First,lets explain what we are talking about. Lets say we have three IP Phones on our network

and we have three phone lines with a dedicated phone number (or DID number) associatedto each phone line. Well how can we configure each analog line & DID to be mapped to oneIP Phone internally for incoming and/or outgoing calling. Here is how we would configurethat on our CME router.

In our configuration example, we will assume we have two IP phones and two analog lines.Phone1 will use extension 201 and Phone2 will use 202. Each phone will be tied to one ofthe analog lines. For external calling Phone1 and Phone2 would need to dial 9 first then thefull number.

STEP 1:TRANSLATION RULESFirst we will configure two translation rules for our two phones. For phone1, if it dials 9 first

then the full number like 1-925-230-2203 then we want to translate the 9 to 19. ForPhone2, another translation is configured where we would translate 9 to 29.

voice translation-rule 1rule 1 /^9/ /19/

voice translation-rule 2rule 1 /^9/ /29/


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STEP 2:TRANSLATION PROFILENext we will associate each of the configured translation rules to their own profile for eachphone under that phones extension in the name. This translation would happen with thenumber we would dial from our IP phone or the called number.

voice translation-profile TP-201

translate called 1

voice translation-profile TP-202translate called 2

Lets quickly discuss calling and called. Calling is when we are specifying the source orthe caller in a phone conversation. Called would be our callee or destination number thatis dialed.

For example, a person at 800-123-4567 is calling someone at 925-230-2203. Our 800number would be our calling or source and our 925 -230-2203 would be our called ordestination.

STEP 3:DIAL PEER (VOIP)Next we will configure two VOIP dial peers for each phone that will associate one of thetranslation profiles configured. Plus we will include what the calling number or answer-address for all calls matching the extension for each phone. So, for example, any numberthat is dialed by phone1 at extension 201 would automatically match that dial peer configured(in our example that would be VOIP dial peer 1). Matching to that dial peer would use thetranslation rules and translate the access code of 9 to 19 if that matches.

dial-peer voice 1 voiptranslation-profile incoming TP-201answer-address 201

dial-peer voice 2 voiptranslation-profile incoming TP-202answer-address 202

STEP 4:DIAL PEER (POTS)Now we would configure two POTS dial peers mapped to an FXO port which has a dedicatedanalog line plugged in. Each dial peer route pattern will have the new translated access code(which can be 19 or 29). So, if phone1 was making an external code and its access codewas translated from 9 to 19 then it would use the configured POTS dial peer 19 routing thecall through that analog port. The T in the translation means any full phone number, but willstrip off the 19 or 29 before going out to the PSTN.

dial-peer voice 19 potsdestination-pattern 19Tport 1/0/0

dial-peer voice 29 potsdestination-pattern 29Tport 1/0/1


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STEP 5:VOICE PORT FOR INCOMINGCALLSSteps 1 to 4 was focused on configuring the phones for outgoing calls through therededicated analog line. Well what about incoming calls being routed to a specific IP phone. Ifa caller calls the user at Phone1 using its external phone number we want that call to beforwarded to Phone1. The configuration is straight-forward by using the PLAR command

followed by the extension we want all calls to be sent to among other necessary configurationincluding enabling caller-ID on the FXO analog port.

voice-port 1/0/0supervisory disconnect dualtone pre-connectpre-dial-delay 0no vadtimeouts call-disconnect 2timeouts wait-release 2connection plar opx 201caller-id enable

voice-port 1/0/1supervisory disconnect dualtone pre-connectpre-dial-delay 0no vadtimeouts call-disconnect 2timeouts wait-release 2connection plar opx 202caller-id enable


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4.6 Configuring FXS port as a SCCP port

By default when analog phones are plugged into analog modules like FXS ports they are notpart of CME or its features such as Hunt Groups. However, there is a way to configure aFXS port that has a connected analog device like a phone to be a SCCP port meaning it canbe part of CME which uses SCCP for all communications with IP phones.

STEP 1:SCCPFirst lets configure globally SCCP to be binded to the BVI10 interface that is being used forCME and phone communication. The IP address on BVI10 is 10.67.78.1

sccp local BVI10sccp ccm 10.67.78.1 identifier 1 priority 1 version 4.1sccp

sccp ccm group 1bind interface BVI10associate ccm 1 priority 1associate profile 1 register mtp001d4567c690

keepalive retries 5switchback method graceful

For the line, mtp001d4567c690, the last part is the mac address from the BVI10 interface.

uc01tra#show interfaces bvI 10BVI10 is up, line protocol is upHardware is BVI, address is 001d.4567.c690(bia 001b.8faa.a860)

STEP 2:STCAPPNext we will enable STCAPP and associate our configured SCCP group to this application.

stcapp ccm-group 1stcapp

STEP 3:CONFIGURE FXSPORT AS A SCCPPORTNext we will configure our FXS port (located on port 0/0/0 on our Cisco UC520 appliance orrouter) to be an SCCP port to be able to communicate with CME. We will also enable caller-ID on this port.

dial-peer voice 14 potsservice stcappport 0/0/0

voice-port 0/0/0caller-id enable


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STEP 4:GET THE MACADDRESS FOR THE FXSPORTNext, get the MAC address associated with port 0/0/0, which will look like this...AN1D4567C690000.

This info can be obtained by issuing the following command:

uc01tra#show stcapp device summaryTotal Devices: 1Total Calls in Progress: 0Total Call Legs in Use: 0

Port Device Device Call Dev Directory DevIdentifier Name State State Type Number Cntl---------- --------------- -------- ------------- ------- ----------- ----0/0/0 AN1D4567C690000 IS IDLE ALG 6776 CME

There we see AN1D4567C690000

STEP 5:ADD MACADDRESS UNDER A NEW EPHONE FOR CMENext we will configure a new ephone for our FXS port with the MAC address we determinedfrom step 4. The type would be ANL and will use the directory number found at ephone-dn10.

ephone 4device-security-mode nonemac-address D456.7C69.0000type anlbutton 1:10

Now our FXS port can place/receive calls including access to other services like voicemail orhunt groups on CME.


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4.7 CME as SIP Server for SIP Clients

Cisco CME router can also be configured to be a SIP server to accept SIP phones on thenetwork working with CME.

STEP 1:CONFIGURE SIPSERVER

First, lets enable our router as a SIP server globally. Most are defaults and our SIP serverwill use the IP address from the router itself listening on the default SIP port number, 5060.Our SIP Server will accept up to 12 directory numbers (or extensions) and 12 phone devices.

voice register globalmode cmesource-address 10.67.78.1 port 5060max-dn 12max-pool 12timezone 47time-format 24date-format YY-M-Ddst start Oct week 8 day Sun time 02:00

dst stop Mar week 8 day Sun time 02:00

STEP 2:DIRECTORY NUMBERSNext we need to configure our directory numbers that would be used for our SIP clients. Inour configuration we will configure two directory numbers, 8701 and 8778.

voice register dn 1number 8701name ROUTEHUB SIP client (X-lite)

voice register dn 2number 8778name Michel Thomatis (SIP)

STEP 3:DEVICE AND DNASSOCIATIONNow we will configure our SIP phone profile to specify the ID or mac address of the SIPclient, the directory number it will use configured from step 2, the codec, and theusername/password that the SIP client will use to authenticate with the SIP server.

voice register pool 1id mac 000C.F179.1682number 1 dn 1username 8701 password cisco6778codec g711ulaw

voice register pool 2id mac 0019.D111.D2E8number 1 dn 2username 8778 password cisco6778codec g711ulaw


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NOTE: Unless this is already configured it is best to configure our CME router toaccept SIP connections with the following configuration among other allowedprotocol communications with H.323.

voice service voipallow-connections h323 to sipallow-connections sip to h323

allow-connections sip to sipsupplementary-service h450.12

STEP 4:USING THE SIPCLIENTOnce the service has been configured, download and install a SIP client (like X-lite) on acomputer. Under the SIP profile settings specify the IP address of the SIP server (our CMErouter), our DN, username, and password. With everything setup the SIP client would beregister to the SIP server router ready to place and receive calls.

4.8 Blocking incoming calls from PSTN

This is a common request for knowing how to block certain phone numbers of calls receivedfrom the PSTN like many telemarketers. Well here is the configuration to block certainnumbers.

In our example, we are blocking 800 number 800-123-4567.

STEP 1:VOICE TRANSLATION RULEFirst we need to configure a translation rule to match and reject the 800 number in questionwe want to block:

voice translation-rule 5rule 1 reject /8001234567/

If we want to block other numbers then those would be added the same as rule 1, but wouldbe added as rule 2 and so forth.


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STEP 2:VOICE TRANSLATION PROFILENext, we will create a voice translation profile called call_blockthat will be map ourconfigured translation rule from Step 1.

Here we are specifying the source or the caller in a phone conversation. This would becalling. Where called would be our callee or destination number that is dialed.

For example, a person at 800-123-4567 is calling someone at 925-230-2203. Our 800number would be our calling or source and our 925 -230-2203 would be our called ordestination. We want to block numbers from certain calling or source numbers into ourvoice network.

This is what our profile configuration will look like:

voice translation-profile call_blocktranslate calling 5

STEP 3:DIAL PEER

Next we will configure a voice POTS dial peer. In our environment we have a single analogline from our PSTN with a single DID number. This analog line is plugged into port 0 on ourFXO module. All calls placed and received are going through this single analog line.

We will apply our call_block translation profile configured from step 2 to our dial peer that isassociated to port 0/1/0 (which is our FXO port connected to our PSTN). This is the samedial peer that is used for placing calls when users internally dial a 9 first then the full number(which is represented as T)for anything local or long distance. All incoming calls cominginto our voice router will match this dial peer because of the syntax incoming called-number.

dial-peer voice 100 potscall-block translation-profile incoming call_block

call-block disconnect-cause incoming call-rejectdestination-pattern 9.Tincoming called-number .port 0/1/0

When that 800 number calls into our voice network it will match this dial peer we configuredand the caller will hear a fast busy because the voice router will reject and disconnect thecall.


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4.9 Setting up an Authenticated SIP Trunk to SIP Provider

Cisco CME router can be configured to form a single authenticated SIP trunk to another SIPcomponent such as another SIP router, server, or even a SIP Provider like ViaTalk. A SIPtrunk can be authenticated (meaning it requires a username and password to be established)or a non-authenticated SIP trunk (no authentication needed and many SIP trunks can be

established). Special call routing, patterns, and translations are needed for placing/receivingcalls through the SIP provider especially if a PSTN provider is connected to the CME routerfor external calling.

STEP 1:AUTHENTICATED SIPCONFIGURATIONFirst we need to configure our authenticated SIP trunk to a SIP provider like ViaTalk. In ourcase, the username would be our dedicated number and the password would be provided tous through the providers control panel. We also need to specify the SIP providers server IPaddress or host/domain name.

sip-uaauthentication username 19252302203 password cisco6778no remote-party-id

retry invite 2retry register 10timers connect 100registrar dns:sanfrancisco-1.vtnoc.net expires 3600sip-server dns:sanfrancisco-1.vtnoc.nethost-registrar

STEP 2:ALLOW SIPCONNECTIONSNext we need to allow SIP connections on our router with other devices that may be SIP orH.323 connections among other details.

voice service voipallow-connections h323 to h323allow-connections h323 to sipallow-connections sip to h323allow-connections sip to sipsupplementary-service h450.12sipregistrar server expires max 3600 min 3600localhost dns:sanfrancisco-1.vtnoc.net

STEP 3:DIRECTORY NUMBER FOR SIPNUMBER

Next we will configure a new ephone directory number that will use extension 7700 and itsfull DID number would be 925-230-2203, our dedicated SIP number.

ephone-dn 13 dual-linenumber 7700 secondary 19252302203name 7700call-forward busy 6000call-forward noan 6000 timeout 15


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STEP 4:TRANSLATION RULE AND PROFILENext, we need to configure two translation rules. In our voice network if we dial 8 first thenthe full number it will be routed across our SIP trunk. If we dial 9 first then the full numberthat would be routed across our PSTN/analog connection.

In our first translation rules any digits that match 8 first then a series of numbers it will strip off

the first digit, which is our access code of 8. Leaving us with the full number that would berouted across the SIP trunk. This translation would be applied for calls that we make fromthe inside.

voice translation-rule 2rule 1 /^8(.*)/ /\1/rule 2 /^8$1[2-9].........$$/ /\1/

voice translation-profile routehub-tp-sip-outgoingtranslate called 2

In our second translation rule any digits or directory number it sees as the source (like 7700,which we configured) will be translated to its full DID number of 925-230-2203. This means

we can place calls across our SIP trunk from other DNs other than 7700.

voice translation-rule 3rule 1 /^.*/ /19252302203/

voice translation-profile routehub-tp-sip-outgoingtranslate calling 3

STEP 5:VOIPDIAL PEERLast we will configure our dial peer required for call routing. This will associate the translationprofile that we configured from step 4 and contain a route pattern with 8 first (our access

code) followed by a pattern numbers that match any local or long distance call. This wouldbe routed across our authenticated SIP trunk (since we specified session target sip -server).We will also include codecs and DTMF details.

voice class codec 1codec preference 1 g711ulawcodec preference 2 g729r8

dial-peer voice 12 voipdescription **Outgoing Call to SIP Trunk**translation-profile outgoing routehub-tp-sip-outgoingdestination-pattern 8[0-1][2-9]..[2-9]......voice-class codec 1voice-class sip dtmf-relay force rtp-ntesession protocol sipv2session target sip-serverdtmf-relay rtp-nteno vad


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STEP 6:PLACING CALLS ACROSS SIPTRUNKTo place calls across our SIP we would dial 8 first (our access code) then the full number.That would match dial-peer 12 that would be routed to our SIP provider. All other callsthrough our PSTN require access code 9 that would be routed through our FXO port.

STEP 7:PHONE REGISTRATION WITH SIPUAThe one thing that occurs with a SIP UA configuration like in our example is that all directorynumbers configured will try to register itself with the authenticated SIP trunk. We only wantour DN of 7700 to be registered with the SIP trunk. Thus, under each DN (ephone-dn) wewill add the command no-reg primary to NOT register with the SIP trunk as a best practice.

In our configuration we will do this for DNs 1001, 1002, and 6700:

ephone-dn 3 dual-linenumber 1001 no-reg primary




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4.10 Phone Directory

You can configure a Phone Directory on our CME router to include a local directory with a listof names pre-configured to be accessed directly from any phone registered with that CMErouter by pressing the Directory button on the phone then choosing Local Directory.

Below is how we can configure names and numbers (extensions or full DID numbers) onCME:

telephony-servicedirectory first-name-firstdirectory entry 1 919252302203 name ROUTEHUB (Main)directory entry 2 912098329950 name Deliver Ease (Main)directory entry 3 912091234567 name Misc Number (Cell)

If the number in one of the entries is external, then the access code (like 9 in ourconfiguration) needs to be included.

Also, any other directory added to CME (SIP or SCCP) will also be included in our localdirectory listing.

As a recap to access the phone directory press the Directory button on the phone thenchoose Local Directory where we can search based on First name, Last name, and/ornumber. Or we can simply press Submit with nothinginputted to display all entries in thedirectory.

4.11 Single Number Reach (SNR)

Single Number Reach (SNR) is a feature that is available with CME version 7 and higher.SNR is a feature that is branded as no more phone tag, which means any person who callsa certain extension enabled for SNR, CME can call another phone number at the same time.

In this configuration, if we dial 1002 (or it's full DID number) it will automatically ring 1001. Ifthere is no answer then the call would go to voicemail (at 6000)

ephone-dn 4 dual-linenumber 1002 no-reg primarymobilitysnr 1001 delay 5 timeout 15 cfwd-noan 6000

Or our SNR number may be an external number like a cell phone or some other externalnumber. However, in our environment to place outgoing calls we need to dial 8 first. This isreflected in our configuration below. Our external number would be our main number forROUTEHUB.

snr 819252302203 delay 2 timeout 30 cfwd-noan 6000


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4.12 Sending Calls to Voicemail (CUE)

This configuration on CME is needed for sending calls to a voicemail system via a SIPconnection particularly to a CUE service engine. But, we want to show you the necessaryconfiguration needed for calls to be sent to voicemail (to Cisco Unity Express) if there is noanswer to a particular directory number.

STEP 1:DIAL PEER CONFIGURATIONFirst we need to configure a dial peer to match our voicemail pilot number (this is the numberwhere we want to send calls to voicemail and/or to access our voicemails). In ourconfiguration that would be DN 6000. This directory number would be our destination patternthat would be forwarded to the CUE service engine found at IP address 192.168.5.2, whichwill be a SIP trunk connection.

dial-peer voice 600 voipdestination-pattern 6000session protocol sipv2session target ipv4:192.168.5.2dtmf-relay sip-notify

codec g711ulawno vad

STEP 2:VOICEMAIL BUTTON CONFIGURATION ON CMENext we will add our voicemail pilot number of 6000 under our CME telephony service. Thisconfiguration will setup a direct speed dial to access our voicemail. When we press theMail button on our phone it will dial the voicemail pilot number.

telephony-servicevoicemail 6000

STEP 3:SENDING CALLS TO VOICEMAIL ON CMEWhen the line (or directory number) is busy or is not answered the call would be forwarded tovoicemail. In our configuration if someone is calling 6700, but the line is busy or there is noanswer (within 15 seconds) then the call would be forwarded to voicemail at directory number6000, which will match the dial peer we configured in step 1.

ephone-dn 10 dual-linenumber 6700 no-reg primarycall-forward busy 6000call-forward noan 6000 timeout 15

STEP 4:CONFIGURE MWINext we need to configure our Message Waiting Indictor (MWI). This means that if a newvoicemail message has arrived, CUE will send a MWI ON message to the number where themessage was left. A red light would turn on the phone. Once the voicemail message is readand no longer new then a MWI OFF message would be sent to turn off the red light.


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In our configuration, our MWI ON directory number will be 8000 and our MWI OFF directorynumber will be 8001. You will also notice that the MWI directory number include . (fourdots) which represents the directory number that is receiving the MWI message. So, if avoicemail message is left for 6700 then the following MWI message is sent: 80006700. Oncethe voicemail message is heard and no longer new then the following MWI is sent:80016700.

ephone-dn 20number 8000.... no-reg primarymwi on

ephone-dn 21number 8001.... no-reg primarymwi off


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4.13 Conferencing

Conferencing is a feature that allows multiple participants or callers to join a single call.Conferencing with a feature called MeetMe allows you to setup conference bridges using aMeetMe directory number on the CME router and allow callers to call directly into theconference bridge.

Ad-hoc conferencing is setup by the user from their IP phone by adding another person to anexisting call. No one can dial into an Ad-hoc conference directly, so its important tounderstand when and how to use these conferencing services on the network.

4.13.1 MeetMe Conferencing

STEP 1:SCCPCONFIGURATIONThis may have been configured already if a feature like enabling an FXS port as a SCCP porthas been configured.

If not configure a SCCP (Cisco Skinny) profile globally binded to the BVI10 interface that is

being used for CME and phone communication. The IP address on BVI10 is 10.67.78.1

sccp local BVI10sccp ccm 10.67.78.1 identifier 1 priority 1 version 4.1sccp

sccp ccm group 1bind interface BVI10associate ccm 1 priority 1associate profile 1 register mtp001d4567c690keepalive retries 5switchback method graceful

For the line, mtp001d4567c690, the last part is the mac address from the BVI10 interface.

uc01tra#show interfaces bvI 10BVI10 is up, line protocol is upHardware is BVI, address is 001d.4567.c690(bia 001b.8faa.a860)

STEP 2:VOICE CLASS CUSTOM TONESNext, configure the tones and frequencies for when callers join a call or leave a call for aconference bridge.

voice class custom-cptone routehub-leave

dualtone conferencefrequency 900 900cadence 150 50 150 50

voice class custom-cptone routehub-joindualtone conferencefrequency 1200 1200cadence 150 50 150 50


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STEP 3:DSPFARMCONFIGURATIONWithin our voice-card on our CME router we need to enable dsp resources (dspfarm). DSPresources are critical for many voice interfaces (like PRI lines) and services like conferencingwithin a Cisco IOS router.

Our DSPFARM for conferencing will include our configured custom tones, the use of the

SCCP application (which is required in order for our conferencing service to interact withCME), the number of conference sessions (in our configuration that would be two), and list asupported codecs on the system.

dspfarm profile 1 conferencecodec g711ulawcodec g711alawcodec g729ar8codec g729abr8codec g729r8codec g729br8maximum sessions 2conference-join custom-cptone routehub-joinconference-leave custom-cptone routehub-leaveassociate application SCCP

STEP 4:ENABLE CONFERENCING UNDER TELEPHONY SERVICESNext we will enable conferencing and our dspfarm profile under our CME service specifyingthat we are using a hardware-based conferencing solution via the DSPFARM configurationwe recently completed.

telephony-servicemax-conferences 8 gain -6sdspfarm conference mute-on 11 mute-off 12sdspfarm units 3sdspfarm tag 1 mtp001d4567c690conference hardware

STEP 5:CONFIGURE MEETME DIRECTORY NUMBERSNext will configure our MeetMe directory numbers to be 6999 and enable conferencemeetme across four ephone-dn profiles for load distribution.

ephone-dn 22 dual-linenumber 6999conference meetmeno huntstop

ephone-dn 23 dual-linenumber 6999conference meetmepreference 1no huntstop


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ephone-dn 24 dual-linenumber 6999conference meetmepreference 2no huntstop

ephone-dn 25 dual-line

number 6999conference meetmepreference 3no huntstop

STEP 6:USING THE CONFERENCE BRIDGE (WITH MEEME)The first thing that needs to happen is the conference bridge needs to be setup first by amoderator or any person who is the lead during a group call. We do this by lifting the phoneoff the hook (any phone within the voice network is fine) then locate the MeetMe softkey onthe phone.

NOTE: If no MeetMe softkey exist then follow the section within this documentdiscussing how to setup a Phone Softkey template.

Pressing the MeetMe softkey you will hear two beeps. There we would input our MeetMenumber of 6999. Moments later the conference bridge will be setup and the moderatorperson would be the first person to join the conference call, which they will see on theirphone as Conference.

Now all other callers can now dial 6999 or the full DID number mapping to 6999 directly tojoin the conference call.

From there any person including the moderator can leave the conference call at any time bysimply hanging up the call.

Another great thing you can do is to view the callers on the conference call by pressing theConfList softkey. From that same list you can also remove any participant from the call asneeded.


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4.13.2 Ad-Hoc Conferencing

The configuration for Ad-Hoc conferencing is similar to the MeetMe conferencing steps 1through 4. Step 5, for our DN numbers will be a little different.

STEP 1:CONFIGUREADHOC DIRECTORY NUMBERSWe will configure our Adhoc directory number to be 6998 and we will enable conference ad-

hoc across two ephone-dn profiles for load distribution.

ephone-dn 26 dual-linenumber 6998name Conferenceconference ad-hocpreference 1no huntstop

ephone-dn 27 dual-linenumber 6998name Conferenceconference ad-hoc

preference 2no huntstop

STEP 2:USING THE CONFERENCE BRIDGE (WITHAD-HOC)Ad-hoc conferencing is a little different to use compared to MeetMe conferencing. In Ad-hocour conference bridge is built and controlled by the user. No caller can call into an Ad-hocconference bridge, they can only be joined into an existing call by the user.

Lets say we have a call setup with someone on the outside. But we want to bring anotherperson into the call. First, we would locate and press the Confrn softkey which will place anew call. This is where we call the second caller we want to join into the existing call. The

first caller will be placed on-hold hearing hold music until the call is completed.

When the call is connected, we can simply press the Confrn softkey again and now we willhave three callers on the same call.

NOTE: If no Confrnsoftkey exist then follow the section within this documentdiscussing how to setup a Phone Softkey template.


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4.14 Paging

Intercom and Paging are two features that work with CME and really operate the same way,but with different results. Paging is a feature that allows you to do announcements orbroadcasts to groups of phones or all phones on the network with only one-waycommunication. Meaning if I setup a Page with a group of people and I speak they cannot

communicate back. This would be good for emergency broadcast messages.

STEP 1:CONFIGURE PAGINGFirst we need to configure Paging and assign what the directory number will be. This canalso be configured or enabled for multicast.

In our example, we will configure Paging to use directory number 6001 plus it will usemulticast address 239.192.2.1 with port number 2000.

ephone-dn 1number 6001name ROUTEHUB Paging Systempaging ip 239.192.2.1 port 2000

STEP 2:ENABLE PAGING UNDER EACH PHONE (EPHONE)Next we will enable paging for each phone with the profile and directory number weconfigured from step 1.

ephone 2paging-dn 1

ephone 5paging-dn 1

STEP 3:USING THE PAGING SYSTEMTo use the paging system from any phone we would dial the DN for paging, which would be6001. This would do a broadcast with to all phone that were associated with that pagingprofile (paging-dn 1). This would be ephone 2 and ephone 5. Remember this is a one waycommunication from one to many.


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4.15 Personal Speed Dial

To setup a personal speed dial or fast dial for a users phone go to that physical ephonedevice on the CME router then add that fast dial including the directory number (or DIDnumber) including the label/name for that number.

Here we will configure a fastdial for number 1001 with a description or label of FR1002 underour IP phone (configured under ephone 6).

ephone 6fastdial 1 1002 name FR1002

To access the Personal Speed Dials go to the Directories button on the phone directly thengo to Personal Speed Dials to access your list of fast dials.

4.16 Upgrading CallManager Express

The best and fastest way to upgrade the Cisco CallManager Express (CME) software versionis upgrade the router (or UC520) IOS itself.

For example, on the Cisco UC520, I was at CME version 4.2

I upgraded to: uc500-advipservicesk9-mz.124-22.YB.bin

Completing the upgrade provided me with CME version 7.1 installed and ready to use withnew features such as Single Number Reach (SNR), which is available within this document.


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4.17 Intercom

Intercom and Paging are two features that work with CME and really operate the same way,but with different results. Intercom is a feature that allows you to do announcements orbroadcasts to groups of phones on the network with two-way communication. Meaning if Isetup an Intercom with a group of people and I speak then they can communicate back.

STEP 1:CONFIGURE INTERCOM DIRECTORY NUMBERSFirst we need to configure two directory numbers for each of the two-way participants.Configuring Intercom directory numbers looks different and includes an A at the beginning.

In our configuration our first Intercom DN will be 5001 (or A5001) that can initiate an Intercomconnection with the second Intercom DN 5002 (or A5002). And vice versa where the secondIntercom DN of 5002 can initiate an Intercom connection with the Intercom DN 5001.

ephone-dn 11number A5001 no-reg primarylabel Intercomname Intercom

intercom A5002

ephone-dn 12number A5002 no-reg primarylabel Intercomname Intercomintercom A5001

STEP 2:ASSOCIATE INTERCOM DNWITH PHONES.Next we will associate the two Intercom DN configured from step 1. One phone will beassociated with Intercom DN 5001 on button #2 for one Cisco 7970 phone. The secondphone will be associated with Intercom DN 5002 on button #2 for another Cisco 7970 phone.

ephone 6type 7970button 1:10 2:113:13 4:3

ephone 2type 7970button 1:10 2:123:4

STEP 3:USING INTERCOMNow everything is configured. To start an Intercom connection from either IP phone simply

press the Intercom line on the phone then we will automatically connect to the other mappedIntercom line configured. The call will automatically be answered and doesnt requiresomeone to answer the Intercom call. Remember its a like broadcast type service.


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4.18 Hunt Group

A Hunt Group is a feature where a single number (or pilot number) is associated with one ormore dedicated phones/extensions. For example, when someone calls support it would bethe pilot number. On the backend, it will call each participant in the hunt group one-by-oneuntil the call is answered. If the call is not answered in a specific amount of time the call maybe routed to voicemail or some other number.

In our configuration example, we will configure our hunt group pilot number to be 6701. Thetwo directory numbers associated with this hunt group will be 6776 and 6700. If there is noanswer within 15 seconds the call is routed to voicemail at 6000.

ephone-hunt 1 sequentialpilot 6701list 6776, 6700final 6000preference 1timeout 15, 15

The hunt group works by a caller dialing 6701 which would proxy in a way to the first DN inthe configuration (if it is not busy), 6776 then to 6700 after 15 seconds.

4.19 Call Park

Call Park is a feature that allows a user to answer a call, place it on PARK (or HOLD really)then retrieve the call from another phone on the network by inputting the Call Park directorynumber.

Call Park is configured under the ephone-dn, and in our example we will configure Call Parkto use directory number 6002 with a timeout of 30 seconds. Meaning we have 30 secondsto retrieve the call once it has been parked.

ephone-dn 2number 6002park-slot timeout 30 limit 10name ROUTEHUB CALL PARK

To use call park, if we receive a call and we want to transfer the call to another phone withinthe network, but we dont know which one. First we would locate the Park softkey andpress it. This will place the caller on-hold and a 30 second timer starts that we need to

retrieve from another phone. On the phone we will see the Call Park number of 6002displayed.

We would go to another phone and dial the call park number of 6002 to retrieve the call.


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4.20 How to setup Phone Softkey templates

If you want to change the layout of the softkey buttons on your phone or add additionalsoftkey buttons for additional services like conferencing or Live record then you need tocreate a template with what the softkey layout will look like. That template would then beadded to the phone(s) and require a reset.

STEP 1:PHONE TEMPLATEFirst, create the phone template you want in terms of the softkeys including the layout of thesoftkeys that would be displayed. The softkey buttons will vary depending on what we aredoing with our phone. For example, the template will differ depending whether a call isplaced on-hold, the phone idle, or if we have a connected call.

Below we will configure our phone template.

ephone-template 1softkeys hold Newcall Resume Select Joinsoftkeys idle Redial Newcall Cfwdall Pickup ConfList Dnd

softkeys seized Redial Pickup Meetme Endcallsoftkeys connected Endcall ConfList Confrn Hold Join Park RmLstC

So if we have a connected call with someone, from our phone we will see End Call,Conference List, Conference, Hold and other softkey buttons available that we can dowith that active call based on the template we setup.

STEP 2:ASSOCIATE PHONE TEMPLATENext step is to associate the phone template under our ephone profile. In our configurationwe will apply this template under one of our Cisco 7970 phones.

ephone 6ephone-template 1type 7970

Once that is done we will need to reset our phone to make our new template become readyto use:

ephone 6reset


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4.21 Extension Mobility

Extension Mobility is a feature, very popular with the Cisco Unified Communication Series,that allows you to login to any IP Phone on the network and your phone profile (consisting ofyour number and speed dials) will be loaded on that phone. This is done by creating a profileand is configured with your directory number and other info. This feature allows users to not

be physically restrained to their phone or location.

STEP 1:EXTENSION MOBILITY PROFILEFirst we need to configure our Extension Mobility profile. We will configure two profiles fortwo phones on the network. They will contain the directory numbers on that phone todayincluding the username/password for authentication.

voice user-profile 1pin 6778user 78 password 78number 6700,A5001,7700,1001 type feature-ring

voice user-profile 2

pin 6700user 70 password 70number 6700,A5002,1002 type feature-ring

This configuration shows two phone profiles. One phone contains directory numbers 6700,7700, and 1001. Another phone within our network has directory numbers 6700 and 1002.

STEP 2:LOGOUT PROFILESThe logout profile is basically identical to what we configured for our profile, but requires adifferent username/password. The logout profile is what happens when a user logs out ofExtension Mobility and the current physical phone resets back to its default.

For example, in our configuration lets say we have two IP phones. One phone (user1) hasthree directory numbers (6700, 7700, and 1001) and our second IP phone (user2) has twolines (6700 and 1002). Two profiles were created globally on the CME router. Lets sayuser1 is at user2s desk. User1 can login to user2s phone to use their own directorynumbers of 6700, 7700, or 1001. When user1 is finished they would need to logout fromuser2 phone, so the phone can be useable for user2. This is where the logout profile comesinto play, which is to restore the physical phone back to its default configuration.

Below we will configure our two logout profiles, but with a different username/password:

voice logout-profile 1pin 6778user 16778 password 6778

number 6700,A5001,7700,1001 type feature-ring

voice logout-profile 2pin 6700user 16700 password 6700number 6700,A5002,1002 type feature-ring


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STEP 3:ASSOCIATE LOGOUT-PROFILE TO PHYSICAL PHONE (EPHONE)Next under each phone (configured under ephone) we will place the logout profile matchingwhat that current phone has in terms of its directory numbers.

ephone 2logout-profile 2

ephone 6logout-profile 1

STEP 4:EXTENSION MOBILITY URLLast we need to configure the phone service that would be used for users to login to theExtension Mobility service that will upload our phone profile.

telephony-serviceurl authentication http://10.67.78.1/voiceview/authentication/authenticate.do

NOTE: the IP address is the IP on our CME router that is associated for our CMEconfigured initially under our telephony-service.

STEP 5:USING EXTENSION MOBILITYThe question now is how do you access the Extension Mobility service to login at any phoneenabled for extension mobility and have our phone profile uploaded.

To login, first, click on the Services button on the phone. Next go to the Extension Mobilityservice listed, login (using the username/password configured under the logout-profile), andthe phone will reload with our authenticated profile. To logout, go back to the ExtensionMobility and click Logout.
http://10.67.78.1/voiceview/authentication/authenticate.dohttp://10.67.78.1/voiceview/authentication/authenticate.do


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4.22 How to setup a custom ring tone?

To create a custom ring tone for your IP Phone complete the following steps.

STEP1:CREATE PCM/RAWAUDIO FILES

First, choose the audio you want such as an mp3 file. This file needs to be converted to a.raw format.

I use the program called Switch by NCH to convert a short playing mp3 file into a raw fileusing 8khz, Mono.

In our example, we have an mp3 file called 24.mp3. This file would be converted to 24.rawusing our NCH Switch application.

STEP 2:CREATE THE XMLFILESNext we need to create two XML files: RingList.xml and DistinctiveRingList.xml

The most common XML file created is the RingList.xml, which will provide a ring tone for ALLlines on a phone. Where the District

vuc-smb-cp-cme

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