CCNA Voice Quick Reference

Michael Valentine

ciscopress.com

Your S h o r t Cut t o K n o w l e d g e

As a final exam preparation tool, the CCNA Voice Quick Reference provides a concise review of all objectives on the new IIUC exam (640-460). This digital Short Cut provides you with detailed, graphical-based information, highlighting only the key topics in cram-style format. With this document as your guide, you will review topics on concepts and commands that apply to Cisco Unified Communications for small and medium-sized businesses. This fact-filled Quick Reference allows you to get all-important information at a glance, helping you focus your study on areas of weakness and enhancing your memory retention of essential exam concepts.

About the Author
Mike Valentine has 13 years of experience in the IT field, specializing in network design and installation. He is currently a Cisco trainer with Skyline Advanced Technology Services and specializes in Cisco Unified Communications, CCNA, and CCNP classes. His accessible, humorous, and effective teaching style has demystified Cisco for hundreds of students since he began teaching in 2002. Mike holds a bachelor of arts degree from the University of British Columbia and currently holds the MCSE: Security, CCNA, CCDA, CCNP, CCVP, IPTX, QoS, CCSI #31461, CIEH, and C T P certifications. He has completed the CCIE written exam. Mike was on the development team for the Cisco Unified Communications Architecture and Design official Cisco courseware and is currently developing custom Unified Communications courseware for Skyline. Mike coauthored the popular CCNA Exam Cram, second edition, first published in December 2005, as well as the third edition of that volume published in December 2007.

A b o u t t h e T e c h n i c a l Editor
Denise Donohue, CCIE No. 9566, is manager of Solutions Engineering for ePlus Technology in Maryland. She is responsible for designing and implementing data and VoIP networks and supporting companies based in the National Capital region. Prior to this role, she was a systems engineer for the data consulting arm of SBC/AT&T. Denise was a Cisco instructor and course director for Global Knowledge and did network consulting for many years.
© 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.

C C N A Voice Q u i c k Reference

by Michael Valentine

Introduction

Introduction
Voice over IP (VoIP) is no longer an interesting sidebar technology; it is a fact of day-to-day life for millions of people, some of whom are not even aware they are using it. Cisco has aggressively pursued the development and deployment of its Unified Communications suite of products and can now offer an integrated voice, video, and data solution for any business, whether it has just a few employees or a hundred thousand worldwide. The technology is reliable, user friendly, and exciting, but it is not simple—and a successful deployment requires that the designers, implementers, and administrators of a Unified Communications system know what they are doing. Training and certification of key staff are strategic components of any business plan to deploy a Unified Communications system. Until recently, the training track for Unified Communications went from the C C N A (the Associate-level routing and switching certification) straight to C C V P , the Professional-level voice certification. The transition between the certifications was difficult for many, because the C C N A did not examine any Unified Communications topics, and the C C V P launched directly into advanced V o I P signaling protocols, Unified Communications Manager administration, traditional telephony, gateway and gatekeeper configuration, Q o S , and so on—all the while assuming that the student had a firm grasp of routing and switching concepts. I have met many good C C N A students who had no telephony or V o I P background and consequently had great difficulty in the C C V P program. Likewise, many students with very strong traditional telephony experience were quickly lost in the intensive data concepts of the C C V P curriculum. It was clear to me and to many of my colleagues that the C C N A was not a good fit as a prerequisite to C C V P . All this brings us to some good decisions that were made regarding Cisco Unified Communications training and certification. The C C N A has itself been split into C C E N T and C C N A , with C C N A serving as the foundation to some new and specialized certifications at the Associate level. The I I U C curriculum prepares students for the C C N A Voice certification, which in turn is a solid preparation for and a much-needed transition to C C V P .

© 2 0 0 8 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.

and some as a refresher right before their test. but you can be sure that the exam will touch on all the topics you find here. too. I hope you find it useful. W h o Should Read T h i s G u i d e Anyone who is preparing to take the CCNA Voice exam will find this guide useful. . or at least fewer dollars wasted. Reviewing this document should help you remember key points and commands you will need to know for the exam. with an unrelenting pressure to be more efficient. Please see page 147 for more details. Efficiencies can be gained by reducing costs. but significant gains can also be made by investing in the business infrastructure so that productivity increases dramatically. to react quickly. Introduction to Unified Communications Today's work environment can be very different from what our parents experienced. Some may use it as in introduction. All rights reserved. Increased productivity means more opportunities to profit from a newfound competitive edge. which in turn increases profit. some perhaps both.CCNA Voice Quick Reference by Michael Valentine Introduction P u r p o s e of T h i s G u i d e This document serves as a roadmap of the CCNA Voice curriculum and a quick reference for the concepts and commands that apply to Cisco Unified Communications for small and medium-size businesses. This publication is protected by copyright. or ROI. © 2008 Cisco Systems Inc. Those of you who are getting back into study mode for a C C V P exam may turn to this guide as a refresher. and to make important decisions instantly. The business environment is more competitive. businesses want to see more dollars earned. welcome and enjoy the text. Data networkers who need a quick but complete introduction to Cisco Unified Communications for a small or medium-size business will find it useful as well. Then there are always those who simply want to learn something new. This document is not a list of all the questions you may be asked on the exam. The goal is to maximize the ROI—for every dollar spent. Whoever you are. This is known as Return on Investment.

The next significant feature of a Unified Communications system is that it is easy to scale. administering. Please see page 147 for more details. and features that are linked by common protocols. The evolution of communications from traditional telephony. and video on a converged single network. and maintaining the network simpler and more cost effective than if three separate systems existed. and support the various components as an integrated system. . Cisco has taken significant steps to develop. and even more features. This publication is protected by copyright. Because the Cisco Unified Communications system is a distributed collection of devices. Unified Communications puts voice. The next section examines the components of a Unified Communications system and introduces the devices and applications that make up the system. document. adding a new component is much simpler. functions. release. Unified Communications also puts powerful applications with information-distribution features right where they are needed. This makes monitoring. adding more users. All rights reserved. and now to Unified Communications. has created opportunities for businesses to access information and get it to workers instantly. data. Workers today can be almost anywhere and can carry out meaningful or even critical tasks anywhere they can get a connection to the converged network. The components required to create and use such a system are numerous and complex. through cell phones. © 2008 Cisco Systems Inc. more locations. to smart phones and email. and integration of the new component's capabilities and features can appear seamless to the people who use the system.CCNA Voice Quick Reference by Michael Valentine Introduction One area in which businesses have found ways to improve their ROI is in their communications.

All rights reserved. Please see page 147 for more details.FIGURE 1 The Unified Communications Architecture © 2008 Cisco Systems Inc. This publication is protected by copyright. .

PCs with software phones. Without QoS. video terminals. This includes Cisco Unified IP Phones. billing systems. and video. One of the critical functions (and one that is unfortunately often underemphasized in many deployments) is quality of service. • Call-Processing Layer: This layer manages the signaling of voice and video calls. routers. or other applications that send and receive information from the Unified Communications system. Because the Unified Communications systems are distributed (meaning not constrained to one box or even one location). Infrastructure design and deployment is literally the foundation of the system. and records the details of the call for future analysis. the applications can be hosted almost anywhere. The call agent carries out many other functions. • Endpoint Layer: This layer includes the parts of the system that the users see. made up of connected switches. . and video between users on the system. The following sections examine the layers in a little more detail. Please see page 147 for more details.• Infrastructure Layer: This layer refers to the network itself. or touch. but with many more features. These include Layer 2 and 3 switches. instructs the phones to play dial tone or to ring. This is the converged network that carries data. voice. When a user picks up the phone and dials a number. call-center applications. routers. the call processing agent determines how to route the call. Voice gateways are among the most important components because they provide the connection to the PSTN or other network carriers. given appropriate connectivity. • Applications Layer: This layer features elements such as voice mail. or otherwise complement the Unified Communications systems. © 2008 Cisco Systems Inc. or QoS. Infrastructure Layer At the infrastructure layer. All rights reserved. QoS provides service guarantees to various types of network traffic. hear. draw from. and voice gateways. voice. in particular voice and video traffic. you can experience poor call quality or even failed calls. This publication is protected by copyright. it can be considered the equivalent of a traditional PBX system. timecard or training systems. and voice gateways. and customer resource management applications—to name just some of the many applications that can integrate with. we are building the connections between all the devices that send and receive data.

they will manifest as system failures or unreliability.if any weaknesses exist here. Transfer. Cisco Unified Communications Manager Business Edition handles up to 500 users and runs as a standalone installation on a 7800-series Media Convergence server. the call agent receives the digits and tries to find a match for the number in its dial plan. The call agent usually keeps detailed records of each call made. All rights reserved. . Call Park. such as Hold. achieving that goal takes careful attention and good design. Cisco Unified Communications Manager can handle 30. The call agent also instructs the phones to tear down the call when one party hangs up. Call Processing Layer The call processing layer is chiefly about the call agents. these are commonly used for billing purposes or troubleshooting. If the destination number is a phone that it controls. the call agent also sets up other services. • • • © 2008 Cisco Systems Inc. Cisco Unified Communications Manager Express serves up to 240 users and runs on the Integrated Services Router platforms. It is very important to build a solid and correct foundation.999% uptime. During the call. Cisco provides several options for call agents. and so on. Please see page 147 for more details. Conference. the call agent instructs the phone to play a dial tone. The system runs on the Cisco Unified Communications 500 Series for Small Business devices. and it determines what to do with the call. When you begin dialing a number to call. A call agent is not a person. it is an application that looks at the signaling traffic from devices that place and receive calls. This publication is protected by copyright. matched to the size and requirements of the customer: • The Cisco Smart Business Communications System is designed for small businesses with up to 48 users. it tells the called device to ring. The goal is to achieve 99.000 or more users and runs on clusters of 7800-series Media Convergence servers. A Unified IP Phone sends a packet to the call agent when you lift the receiver.

This publication is protected by copyright. and the call agent software can support up to 48 phones. firewall. Voice mail and Auto-Attendant functions are provided by the integrated Cisco Unity Express application. intrusion prevention. All rights reserved. It is essentially a solution-in-a-box.Smart Business Communications System FIGURE 2 The Smart Business Communications System—Image © Cisco The Smart Business Communications System is a group of specially designed. providing the kind of connectivity options small businesses need to allow them to take advantage of Unified Communications with a good ROI. integrated devices that can provide highquality routing. Please see page 147 for more details. The Unified Communications 500 Series devices are small and inexpensive. © 2008 Cisco Systems Inc. wireless. . with a simple web-based interface that is largely plug and play. Power over Ethernet. The SBCS is expandable using 500-series switches. and many WAN and PSTN connectivity options.

The call agent application is embedded with the Cisco IOS software and is configured either from the command line or a Web-based interface. All rights reserved. The Unified CM Express system can be set up either as a PBX or a Key switch system. 1800. site-to-site connections are possible in a variety of environments. 3800. 2800. and 7200-series platforms. and scalable and integrates with both Service Provider connections and Unified Communications Manager clusters.a. . providing customers with a familiar experience that suits their operating environment. Unified Communications Manager. Unified CM Express is a fullfeatured call agent that is cost-effective. Business Edition is a standalone installation of the Unified CM application and Cisco Unity Connection. With support for both H. coresident in a single M C S 7800-series appliance. This system can support up to 500 users in a single site or multisite centralized deployment and can be migrated to a full CM cluster if growth necessitates it. reliable.k. including the 800. Unified CM Business Edition provides medium-size businesses with advanced features such as Mobility (a. Please see page 147 for more details. Single Number © 2008 Cisco Systems Inc. Business Edition Unified Communications Manager. This publication is protected by copyright.323 and SIP protocols.Unified Communication Manager Express FIGURE 3 Cisco Integrated Services Routers for Unified Communications Manager Express— Image © Cisco Cisco Unified Communication Manager Express is a software feature that can run on the ISR-series router platforms.

with the capability to form intercluster connections. because third-party applications can be developed to closely integrate with the Cisco suite of products. Please see page 147 for more details. Intercom Whisper. . Because Unified CM Business Edition uses the same call agent software as a full cluster deployment of Unified C M . Unity Connection.Reach). which in turn allows 911 responders to locate the device (and therefore presumably the emergency) more precisely. Unified Communications Manager The full version of Unified Communications Manager is an enterprise-class.x are Windows based. Contact Center Express. This information is attached to the caller information in the event the device calls 9 1 1 . Do Not Disturb. as well as speech recognition and integrated messaging. Unified Personal Communicator. The following is a list of the more common applications found in a Unified Communications system: • Voice Mail: Voice mail can be provided using Cisco Unity.x are Linux-based appliances. The maximum mailboxes and recording time capacities vary depending on which module (either Advanced Integration Module or Network Module) is installed in the router. 911 operation in a Unified Communications environment is a major design challenge because a VoIP phone system can easily throw out the premise that a PSTN call is placed from the same location as the phone that made it.000 users per cluster. whereas versions 5. and Unity Express is a self-contained module that is added to an ISR router and administered through the command line and GUI. Unity and Unity Connection run on the MCS 7800 series platforms. or Unity Express. All rights reserved. Cisco Emergency Responder: This application tracks the location of an IP telephony device based on the physical switch port it is connected to. This publication is protected by copyright. and so on. such as Unified Presence. it supports full integration with the other Unified Communications applications. • © 2008 Cisco Systems Inc. fully scalable. redundant. it can support a global unified communications system for hundreds of thousands of endpoints.x and 6. and Audible Message Waiting Indication. Unified CM versions prior to 5. Applications Layer There are effectively a limitless number of applications that can be part of a Unified Communications system. MeetingPlace Express. and robust distributed packet-telephony application. Scalable to 30.

speakerphone. Please see page 147 for more details. The following is a partial list and brief description of the Cisco Unified IP Phones available: Commercial/Retail Phones 7931G: 24 programmable buttons. whiteboarding.729 codecs (and. Versions are available to support small and large call centers. escalation and logging. All Cisco Unified IP Phones provide a display-based user interface. All rights reserved.x+ to indicate presence information. user customization. The native capability includes on/off hook status in speed dials and call lists." "Do Not Disturb. Cisco Unified Presence: This extends the native capabilities of Unified CM 6. XML-PTT. 4-way LEDs. and support for G. 2-in. whereas the full applications server provides detailed presence information as typically found in chat applications ("On the Phone. color screen. • • Endpoints Layer An increasing variety of Cisco Unified IP Phones (and third-party IP phones) can be part of a Unified Communications deployment. Cisco Wideband and/or iLBC codecs). Power over Ethernet capability (where appropriate). emulates 7970G functionality © 2008 Cisco Systems Inc.711 and G. longer battery life IP Communicator: Software-based IP Phone. speakerphone. color screen. .• Cisco Unified Contact Center [Express]: This is a call center application with full feature support for advanced call distribution. Dedicated HolaVTransfer/Redial buttons 7921G: Wireless." "Out to Lunch." and so on). and conference participant management. This publication is protected by copyright. supervision. Cisco Unified Meeting Place [Express]: This is a full-featured web-conferencing application enabling voice and video conferencing as well as document sharing and collaboration. XML-PTT. on some models. 2-in. longer battery life Mobility 7921G: Wireless.

XML-capable. XML-capable. XML-capable. XML-capable. 2-button. XML. SIP-capable 7975G: Hi-fidelity audio. 6-button. Hi-fidelity audio. SIP-capable 7965G: Gig Ethernet. XML-capable. 6-button. This publication is protected by copyright. 3 softkeys. Hi-fidelity audio. SIP-capable 7971G-GE: Gig Ethernet. XML-capable. 2-button. Hi-res color display. 8-button. SIP-capable 7975G: Gig Ethernet. Backlit hi-res color touch screen.Business Class 7940G: B/W LCD. SIP-capable Advanced Media 7942G: Hi-fidelity audio. 6-button. XML-capable. XML-capable. extended audio coverage w/ extra mics. XML-capable. 6-button. SIP-capable Video 7985G: Personal desktop video phone Unified Video Advantage: Software IP Video Phone with support for attached camera Conference 7936G: Backlit LCD. backlit hi-res color display. 2-button. SIP-capable 7961G: Higher resolution B/W LCD. SIP-capable 7945G: Gig Ethernet. Hi-res display. small-medium conference needs 7937G: Hi-fidelity audio. SIP-capable 7960G: B/W LCD. Hi-res display. Please see page 147 for more details. 8-button. All rights reserved. Hi-res color display. SIP-capable Color Touch 7970G: Backlit hi-res color touch screen. 2-button. Backlit hi-res color touch screen. SIP-capable 7941G: Higher resolution B/W LCD. . XML-capable. SIP-capable 7962G: Hi-fidelity audio. Hi-fidelity audio. XML-capable. 6-button. 8-button. large display © 2008 Cisco Systems Inc.

Interactive Voice Response (IVR). we examine the variety of applications available for integration in a Unified Communications environment. This publication is protected by copyright. . and Presence. AIM modules are connected to the main board as a daughter board addition and use flash memory for greetings and message storage. Mobility.Understanding Unified Communications Applications In this section. NM modules are inserted into module bays in ISR routers. Cisco Unity Express Unity Express is an ISR-based application that runs either on an AIM module or an NM module. All rights reserved. Auto Attendant. Contact Center. Please see page 147 for more details. Users Messaging Capability Platform T D M PBX Integration? Networking? Redundancy? Unity Express Unity Connection Unity 250 3000 7500 per server Voice Mail + Integrated Messaging Voice Mail + Integrated Messaging Voice Mail + Integrated Messaging + Unified Messaging ISR MCS MCS No Yes Yes Yes No Yes No No Yes The following sections describe the messaging products listed in the table in more detail. AIM modules therefore have less capacity for storage. including Messaging. Product Max. The following table provides a summary of the options. Messaging A variety of messaging options are available to suit the needs of businesses small and large. use a hard © 2008 Cisco Systems Inc.

Text-to-speech capability allows users to have their emails read to them over the phone by the RealSpeech engine. Users can define their own rules to transfer calls based on caller. voice mail. and GroupWise. Unity Express is managed through the command line or a web-based GUI. Full unified messaging is possible with connectors for Exchange. and possible from almost anywhere. search. email application. or record messages hands-free.000 users in a multi server networked environment. and have greater capacity for storage than A I M modules. Unity Connection also supports speech recognition. This publication is protected by copyright. or Cisco Unified Personal Communicator. and fax messages. Cisco Unity Connection Unity Connection is a medium-size business solution with a full range of messaging features. It can be deployed on its own or as a coresident installation as part of Unified Communications Manager Business Edition on suitable MCS platforms. . Access to messages is made simple. Unity supports 35 languages. a web GUI.disk for greeting and message storage. Please see page 147 for more details. All rights reserved. Unity connection supports up to 3000 users per server (dependent on hardware). When deployed as part of CM Business Edition. allowing encrypted messages and preventing messages that have expired from being played. Scalability is achieved by networking up to 10 other Unity messaging products of any type. speech recognition is also available so users can instruct Unity to play. Unity Connection supports up to 500 users. Secure messaging is supported. time of day. notably Octel and Nortel systems. Interoperability with legacy voice-mail systems. Notes. and Microsoft Exchange calendar status. Unity Express supports from 4 to 16 concurrent sessions and 12 to 250 mailboxes (dependent on the module and platform installed). It allows users to view and sort their voice messages using the IP Phone display. Fourteen languages are supported for deployments worldwide. or IMAP client. providing a single inbox for email. facilitating deployments worldwide. when deployed as a standalone application. allows a phased transition to IP messaging with minimal disruption to users. intuitive. Cisco Unity Unity is the enterprise-class messaging application with support for up to 7500 users per server and up to 250. an email client. Multiple interfaces are supported for managing messages from an IP Phone. allowing users to speak commands to manage their messages hands-free. Unity Express can be deployed in conjunction with Unified CM or CM Express and can supplement a full Unity deployment. © 2008 Cisco Systems Inc.

and Unity Express all provide Auto Attendant functionality. which number was called. and often they allow the caller to spell out a first or last name to search in the company directory. and reporting features. © 2008 Cisco Systems Inc. . what numbers the callers pressed in response to the greeting they heard. database integration. their functionality is limited to pretty basic menu navigation. If you have ever heard: "For service in English. Call centers that have a high call volume and many possible queues of callers waiting for different agent capabilities can effectively deploy Unified IP IVR to steer callers to the correct agent. and Java application integration. followed by the # sign"). Unity Connection. Unified IP IVR includes the capability to provide both real-time and historical reports on its utilization and offers multiple-language support. it can play several. including speech recognition. Typically. management. Cisco Unified IP IVR has all these advanced capabilities.Auto Attendant An Auto Attendant is basically an advanced answering machine. and powerful call routing. prompt-and-collect ("Please enter your 10-digit account number. Cisco Unity. appuyez sur le 2 . Cisco Unified Customer Voice Portal For the very largest call centers. the Unified CVP product provides advanced IVR. Cisco Unified IP IVR Although Auto Attendants are useful. integration with Cisco Unified Contact Center (Enterprise and Hosted). instead of only one message. Please see page 147 for more details. press I. Auto Attendants allow callers to select the department or extension they want to call. " you have been served by an Auto Attendant. Unity and Unity Connection include a simple web interface that makes it very easy to construct menus and test to see that they work as you intended. This publication is protected by copyright. advanced queuing. a much more advanced IVR application is required. Text-to-Speech. . or perhaps to an automated information source without the need to tie up an agent at all. and most importantly. . Pour service en Francais. depending on the date and time. To scale this functionality up to call-center size. All rights reserved. and especially to include speech recognition. .

The desire to have a seamless transition between the various ways in which people can be reached has spurred the development of mobility features in Cisco Unified Communications. Contact Center Hosted provides all the advanced capabilities found in Contact Center Enterprise. and even video interaction. Subscribing business customers can have IP or time-division multiplexing (TDM) infrastructures or a combination of the two. and chat and web collaboration in a singleserver.k. it provides sophisticated call routing. and utilizes multiple technologies to communicate. outbound dialing capabilities.a Single Number Reach) Allows multiple remote destinations (commonly a cell phone." Cisco Unified Contact Center Enterprise: Provides intelligent contact routing. Enterprise. • • Cisco Unified Mobile Solutions Today's workforce is mobile. or other work location) to be configured to ring at the same time as the worker's enterprise © 2008 Cisco Systems Inc. Sophisticated monitoring allows customers to be routed to the most appropriate agent (based on real-time conditions such as agent skills. The three Contact Center products are described next: • Cisco Unified Contact Center Express: Suitable for 10 to 300 agents. It combines multichannel automatic call distributor (ACD) functionality. and multichannel contact management. distributed. integrated "contact center in a box. The key products are the following: • Cisco Unified Mobility: (a. including basic telephony as well as feature-rich web. availability. who then lease its functionality to customers who want a virtual contact center without the need to manage and maintain it themselves. network-to-desktop computer telephony integration (CTI). Customer contact solutions provide multiple avenues to reach and interact with customers. All rights reserved.Cisco Unified Contact Center Cisco provides a range of Contact Center products for S M B . Please see page 147 for more details. and queue lengths) anywhere in the enterprise. This publication is protected by copyright. comprehensive contact management. regardless of the agent's location. a home office phone. call treatment. and Service Provider applications. email. . Cisco Unified Contact Center Hosted: An application hosted by service providers.

when a customer calls your work number while you are on your way to a meeting. Integration with an Outlook toolbar provides click-to-call or click-to-chat from a message or contact. presence indicators. Presence indications ("Busy. called Cisco Mobile Voice Access." "In a call. Cisco Unified IP Communicator: A fully functioned software IP Phone. By dialing a configured number and entering an access code.desk phone. Cisco Unified Presence: A server-based application that extends the on/off hook status monitoring capability of Unified CM 6. ." "Do Not Disturb. Status indications can be displayed or integrated with Personal Communicator. voice-mail access. and IBM Sametime Communicator. Furthermore. and collaboration and conferencing integration with Unified Meeting Place. the Microsoft Office Connector. • Cisco Unified Personal Communicator: A desktop PC (or Mac) application that combines a software IP Phone. This is useful not only for presenting the preferred Caller-ID number to the customer. video. A related feature." and so on) can save time and enhance productivity because users can see the status of the person they want to contact before trying to reach them.x to include IM-like status messages. Thus. This is typically achieved through a VPN connection. and online collaboration capabilities. IM client. you can simply pick up your desk phone and continue the call. IP Phone Messenger. All rights reserved. allows users to place calls from their enterprise desk phone from a remote location or a cell phone. which integrates a PC webcam for video calls. your cell phone can ring and you can answer without the customer realizing you are away from your desk. secure text/chat. This publication is protected by copyright. Please see page 147 for more details. the enterprise system will prompt for the number you want to call. Unified IP Communicator can be enhanced with Unified Video Advantage. often characterized as a "7970 under glass." "Away. Cisco Unified Mobile Communicator: An application for smart mobile phones that provides access to enterprise directories. but also potentially for long-distance toll savings. call history of any of the user's phones displayed on the mobile handset. Mobile Communicator. it is perfectly possible to place a call from an airport boarding lounge or your local coffee shop." Users can place and receive calls from their PCs from anywhere that connectivity to the call agent can be established. • • • © 2008 Cisco Systems Inc. and the call will be placed as if you were at your desk. if you return to your desk.

Please see page 147 for more details. with commensurate demands on bandwidth. up to 36 locations can be included in a single conference with nearzero latency. In combination with the Telepresence Multipoint Switch.nified Communications Applications Cisco Telepresence Cisco Telepresence is a state-of-the-art high-definition videoconferencing system. . FIGURE 4 The Cisco Telepresence 3000 System—Image © Cisco © 2008 Cisco Systems Inc. A specially designed system of furniture. All rights reserved. CD-quality spatial audio. This publication is protected by copyright. and high-quality lighting. and microphones creates a life-sized illusion of a meeting whose participants may be half a world apart. This can only be described as a high-end solution. monitors. With 1080p HD video. the experience is dramatic to say the least. cameras.

The CO switch is programmed so that it knows which phone number (subscriber line) is attached to a particular port. This publication is protected by copyright. Please see page 147 for more details. © 2008 Cisco Systems Inc. concepts. Telephone numbering plans are organized so that calls are routed efficiently through the switch system to the correct destination switch. The PSTN FIGURE 5 Public Switched Telephone Network A Representation of the Public Switched Telephone Network (PSTN) The PSTN. All rights reserved. . the call is routed over an interoffice trunk to another switch. or Public Switched Telephone Network. which may have the called subscriber line connected directly to it or may in turn route the call to other CO switches. If the number called is not on the local switch.Understanding Traditional Telephony This section introduces traditional telephony systems. is made up of Central Office switches to which subscriber lines are connected. and applications.

Business Telephony Systems Businesses have more elaborate requirements of the telephone beyond simply placing calls. and trunk cards to © 2008 Cisco Systems Inc. This publication is protected by copyright. As a business grows. line cards to connect to phones. Please see page 147 for more details.Note that for our purposes. a terminal interface that connects phones to the features they want to use. and suppliers. and external calls are routed over a CO trunk to the PSTN CO switch if the called number is not on the PBX.or 5digit numbers) instead of PSTN calls. A PBX consists of a control plane (the "brain"). usually located in their building. a line connects to a single phone number and supports one call at a time. Both have their place. a switching engine that determines which port to route a call out. The PBX is configured in much the same way as the PSTN CO switch: it holds the dial plan for all numbers within the business. whereas a trunk interconnects two switches and supports multiple calls at a time. PBX Systems FIGURE 6 A Representation of a PBX System Business telephone systems often use a Private Branch Exchange (PBX) switch. . and both offer calling features that make it easier to carry on business both internally and externally with staff. customers. two main types of business systems have evolved: the PBX and the Key System. All rights reserved. it is common to install another PBX in another location or building and set up a special trunk (called a tie-line or tie-trunk) between the PBXs so that calls to the remote location are still internal numbers (typically 4. Over time.

Telephony Signaling Telephony signaling refers to the messages that must be sent to set up and tear down a phone call—that is. Following are the three types of telephony signaling: • Supervisory: Communicates the current state of the telephony device. the switch sends a dial tone to indicate that it is ready to receive digits. that is. Older systems also support pulse dialing. The other end of the call hears a ringback tone. with additional advanced features optional based on hardware capability and licensing. supporting from 10 to 20. this is the same as being off-hook. The combination of tones tells the switch what number was pressed. Conference. and so forth. (Note that if the speakerphone button is pressed. • Address: Communicates the digits that were dialed. Address signaling is most commonly done using Dual Tone Multi Frequency (DTMF) tones. This publication is protected by copyright. PBXs come in a variety of sizes. . Only the ringer is active in this state. and whoever is able to pick up Line 2 (for example) can push the Line 2 button on any phone and take the call. These features typically include Hold. This signals the phone switch (PSTN. All PBXs offer basic calling features. anything other than the actual voice. Please see page 147 for more details. alerting the user that there is an inbound call. Voice Mail. Transfer. but key systems tend to have fewer features than PBXs. PBX. One characteristic of key systems that many businesses specifically request is distributed answering from any phone. which is what the old-fashioned rotary dial © 2008 Cisco Systems Inc. All rights reserved. all the phones ring at once. There are three types of supervisory signals: • On-Hook: The phone is hung up.) Off-Hook: The phone receiver is out of the cradle. A key system is like a PBX in that it controls a group of local phones.000 phones. or Key) that the phone wants to make a call. commonly known as TouchTone dialing. PBXs don't normally do this. Park.CCNA Voice Quick Reference by Michael Valentine Understanding Traditional Telephony connect to the PSTN or to tie trunks to other PBXs. they have a central answering point (a receptionist or Auto Attendant) and Direct Inward Dial numbers (DIDs) if needed. • • Ringing: The switch sends voltage to the phone to make it ring. Key Systems Smaller businesses will sometimes use a key system.

Pulse dialing works by repeatedly opening and closing the circuit to the phone switch. The capabilities of SS7 have allowed the introduction of relatively complex value-added services. These tones. for example. and even between national telephone providers in other countries. and others not mentioned here.CCNA Voice Quick Reference by Michael Valentine Understanding Traditional Telephony phones used. SS7's primary role is to complete the setup and teardown of phone calls. Informational signals include dial tone. . and reorder tone. and prepaid calling cards. Please see page 147 for more details. The following steps refer to Figure 7. PSTN Call Setup To make a PSTN call. ringback tone. the switch counts the number of pulses and interprets that as the number dialed. In fact. several steps occur that the caller is unaware of. All rights reserved. You might have seen in really old movies when someone picks up the phone and taps the receiver cradle repeatedly. this is quite a distinct process from the actual transport of the voice signal. This publication is protected by copyright. the call control information in an SS7 network must traverse an entirely separate network from the voice path. between long-distance carriers. ringback tone sounds very different from what would be heard in North America. • Informational: Communicates the call status to participants in the call. FIGURE 7 PSTN Call Setup 0 Customer Telephone © 2008 Cisco Systems Inc. such as call screening. this was how you got the attention of the operator. In England. number portability. will vary from country to country. Signaling System 7 (SS7) SS7 is a global telephony standard that allows a phone call to be routed between CO switches.

The fact that all this happens with very high reliability billions of times every day is pretty impressive. A numbering plan is an organized distribution of telephone numbers administered by a regional or national authority. 8. followed by a three-digit area code. All rights reserved. It also provides some insight into how complex it is to duplicate these functions in a VoIP system. The local CO switch detects that current is flowing over the closed circuit and sends a dial tone to the calling phone. . The ringback tone is heard at the calling party end. and a voice circuit is established end-to-end. © 2008 Cisco Systems Inc. the North American Numbering Plan defines a country code of 1. The called party goes off-hook. Supervisory signaling indicates to the far-end trunk that a call is inbound. 4. The calling phone goes off-hook. Address signals (DTMF or pulse) are sent as the calling party dials the called number. The plan defines the rules that allocate numbers according to an established international telecommunications standard. Local numbers must always be dialed.1. it uses an SS7 lookup to locate the destination CO switch. Numbering Plans NOTE Codes do not always need to be dialed. This publication is protected by copyright. 6. Please see page 147 for more details. 3. closing the circuit to the local CO switch. More on that later. 7. 2. a threedigit office code. There are numerous other numbering plans for other countries or regions of the world. For example. 5. and a four-digit local number. The PBX determines which internal line the call should go to and causes the connected phone to ring. The local CO switch collects the digits and makes its routing decision. in this example.

generating over 100 trillion unique strings. which is used in film and TV. When Tommy Tutone recorded "867-5309/Jenny. Several ranges are also reserved for Easily Recognizable Codes (ERCs). and the " X " represents any digit 0 through 9. or education. Please see page 147 for more details.164 Addressing The E. and 911 numbers that are not used as area codes but for other special assignments. One commonly recognized one is 55501XX. In theory. demonstrations. It is very important to note that the " N " represents any digit in the range 2 through 9. the 3-digit office code. 6 1 1 . and the 4-digit local number. these are numbers where the second and third numbers of the area code are the same.164 addressing scheme is an international standard for telephone numbering plans. as shown here: NXX-NXX-XXXX NOTE Several other ranges are reserved for specialized purposes." he immediately annoyed thousands of phone customers worldwide. © 2008 Cisco Systems Inc. An E. Another recognizable assignment is the " N i l " series: this includes 4 1 1 . those numbers are either reserved for specialized purposes or would interfere with things like operator access numbers. so calling a number seen in a movie will not pose a nuisance to anyone. No actual customer is assigned these numbers. They are used for special services—for example.The North American Numbering Plan Let's look at the N A N P in more detail. All rights reserved. 877. This publication is protected by copyright. it's possible to direct dial any conventional phone in the world from any other conventional phone. originally developed by the International Telecommunication Union. and 866 are toll-free numbers.164 number is standardized at 15 digits. . 800. You will never find an office or area code of OXX or 1XX. 888.164 number contains the following components: CC—Country Code NDC—National Designation Code SN—Subscriber Number An E. E. such as information or emergency services. The 10-digit number is made up of the 3-digit area code.

Alternatively. Please see page 147 for more details. The switch or FXS gateway port must provide power.Introduction to Analog Circuits Analog (in contrast to digital) circuits are still the most common telephone connections worldwide. the analog circuits that Cisco supports are Foreign Exchange Station (FXS). and dial tone to the analog device. although more and more digital phone services are being installed. and Earth and Magneto (E&M). © 2008 Cisco Systems Inc. The phone line to a North American home is most commonly an analog loop circuit. This section examines the components of an analog telephone and the signaling methods used by analog circuits. call progress tones. Components of an Analog Phone An analog phone includes the following components: • • Receiver: The handset speaker Transmitter: The handset microphone • 2-wire/4-wire hybrid: Converts 2-wire from the CO to 4-wire in the phone • Dialer (tone/pulse): The dialing keypad or rotary dial • • Switch hook: The switch that closes/opens the circuit (off-hook/on-hook) Ringer: Sounds to indicate inbound call Foreign Exchange Station An FXS port connects directly to an analog phone or fax machine. Switches (including CO switches and PBXs) and Cisco gateways will have FXS ports to connect an analog phone. This publication is protected by copyright. . a Cisco Analog Telephony Adapter can serve as a remote FXS-to-Ethernet converter to connect an analog station to the VoIP network. All rights reserved. Foreign Exchange Office (FXO). An FXS port on a gateway is also the direct connection to the VoIP network and consequently also contains a coder-decoder (Codec) to convert the analog signal to digital for packetization. Cisco gateways must connect to various analog services to place calls to the PSTN.

This publication is protected by copyright. If you want to connect your gateway router to the phone company over standard analog lines (that you could plug your analog phone into). These ports allow the gateway to place and receive calls to/from the PSTN. it is seldom seen on trunk connections. . A local loop is a two-wire service that uses very simple electrical signaling.Foreign Exchange Office An F X O port connects to the PSTN CO switch. All rights reserved. remember that this technology has been in use and substantially unchanged for 100 years! © 2008 Cisco Systems Inc. Please see page 147 for more details. you use F X O ports. FIGURE 8 Loop-Start Signaling Loop-start signaling is commonly associated with local loop circuits (such as an analog line to the PSTN). FXO ports also include a codec.

the CO switch applies 90V AC current to the open circuit. . it is very rare to see a ground-start trunk in a VoIP network or indeed in any new trunk deployment. The advantage is that it makes glare much less likely. Loop-start works very well for homes or other lightly used circuits. by coincidence. the circuit closes and electricity flows. This is achieved by both ends being able to ground one of the wires in the circuit. This publication is protected by copyright. 2. By the way. All rights reserved. When the receiver is lifted. both ends of the circuit have the capability to detect current. These terms date back to the use of 1/4-inch jacks with a positive contact at the tip and a negative conductor in the ring. The CO switch that is connected to the local loop detects the current flow and interprets this as an attempt to place a call—we say "seize a circuit. but if it is in constant use. we say opens the circuit. the current can be applied even on the open circuit. The DC voltage won't do much. However. Ground-Start Signaling Ground-start signaling is an adaptation of loop-start. These wires (or leads) are referred to as Tip and Ring. because it is AC.Following is the loop-start process: 1. This current is . No electricity can flow because of the open circuit.4 8 V DC. Instead of the circuit being closed only at the phone end." The CO switch plays dial tone down the line to the phone as an indication that it is prepared to collect digits. 3. so that you pick up the phone and there is a caller on the line at the same moment. and consequently ground-start is appropriate for trunk connections that are heavily used. © 2008 Cisco Systems Inc. If the phone is on-hook and the CO switch has a call inbound for it. and both ends can request and confirm the use of the circuit. A phone that is on-hook breaks the electrical circuit. a problem known as glare can occur. this refers to both ends of the circuit being seized at the same time. this is why you should not have an analog phone near the bath. Please see page 147 for more details. but you will definitely know it if the phone rings and you get zapped by the AC voltage.

FIGURE 9 Ground-Start Signaling The ground-start process as it occurs on a trunk between a PBX and the CO switch is described next. The PBX senses the tip ground and closes the loop between tip and ring in response. 2. All rights reserved. The CO senses the ring lead as grounded and grounds the tip lead to signal the PBX that it is ready to receive the call. refer to the diagram for each step: 1. . and communication can begin. It signals to the CO switch that there is an inbound call by grounding the ring lead. The voice circuit is complete. 3. The PBX has a call that it must send to the PSTN. © 2008 Cisco Systems Inc. Please see page 147 for more details. the PBX also removes the ring ground. 4. This publication is protected by copyright.

Types II and V can be connected back-to-back and Type I cannot be. This publication is protected by copyright. (This time period is the delay dial signal. When the far-end device answers the call. even though it has sent the wink. Five types of E & M signaling exist. numbered Type I through Type V. one side is called the trunk side. In an E & M connection. a PBX) has gone off-hook. this is the CO. conversely. waits a set time (perhaps 200ms). it then goes on-hook. the delay compensates for this. a Cisco Gateway) uses a brief off-hook-on-hook "wink" to acknowledge that the originating side (for example. Upon receipt of the wink. The other side is called the signaling-unit side." and "Earth and Magneto. All rights reserved. Three main techniques are employed in E & M circuit signaling: • Wink Start: The terminating side (for example.E&M Signaling Variously called "Ear and Mouth. the CO goes off-hook (called Answer Supervision). The advantage of Delay Dial is that some equipment is not ready to receive digits instantly.) The PBX sends digits. Please see page 147 for more details. and then begins sending digits whether or not the terminating side is ready. . When the far-end device answers the call. • Immediate Start: The originating side goes off-hook. or Cisco gateway E & M interface. E & M connections have separate leads for signaling and voice. © 2008 Cisco Systems Inc. the originating side begins sending digits. The CO then goes off-hook until it is ready to receive digits." E & M analog trunks were typically used to interconnect PBXs (tie-trunks). the terminating side goes off-hook and the voice circuit is then set up. the PBX goes off-hook. the M lead is used to indicate to the signaling-unit side that the trunk side has gone off-hook. The E lead is used to indicate to the trunk side that the signaling-unit side has gone off-hook. and the voice circuit is then set up. this is usually the PBX side. • Delay Dial: Assume that a PBX is placing a call outbound to the PSTN: First. Cisco does not support Type IV. In a Cisco Gateway application. channel-bank." "RecEive and TransMit. the signaling leads are known as the E and M leads.

All rights reserved. the conversion from analog to digital is performed by a codec. we are assuming the calls are not compressed. There are two main types of digital circuits: Common Channel Signaling (CCS) and Channel Associated Signaling (CAS). Because the highest frequency in human speech that we want to reproduce in telephony is around 4000 Hz. The following sections discuss the conversion of analog to digital. more on this later). but the amount of digital data produced is much larger. Please see page 147 for more details. Digitizing Analog Signals There are four steps in the process of digitizing analog sound: 1. an analog circuit can handle only one call at a time. which must encode both much higher and much lower frequencies. A PRI Tl can support 23 calls. Optionally compress the sample Sampling could be done any number of times per second. (For these values. the higher the audio quality. the more samples taken per second. © 2008 Cisco Systems Inc. the sampling rate for standard tollquality digital voice is 8000 intervals per second. CCS circuits are designated as PRI T l . samples at about 192. Sample the analog sound at regular intervals 2. Encode the value into a binary expression 4. Nyquist's theorem states that the sampling interval should be 2x the highest frequency of the sample to produce acceptable audio quality during playback.Introduction to Digital Circuits Digital circuits have the chief advantage of allowing a much higher density of calls on a given physical connection. a PRI El 30.048Mbs supports 30 calls. Quantize the sample 3.000 times per second. and a BRI only 2. and BRI. PRI E l . The use of a digital circuit by definition implies that the voice signal must be digitized. CD music audio. . and El at 2.544Mbs supports 24 calls. whereas a digital circuit can handle many. CAS circuits are available in two speeds: Tl at 1. By contrast. This publication is protected by copyright.

Quantizing refers to making a digital approximation of an analog waveform. Imagine drawing an arc on a chessboard; if you had to define the arc using only the square it was in for each row (segment) and column (interval), you would end up with a stepped pattern that was sort of close to the original arc but not exact. This is exactly the process that happens with quantization: the codec chooses a segment value that is as close as possible to the analog value at the interval it was sampled, but it cannot be exact. To make the quantization more accurate, each sample is divided into 16 intervals that are adjusted to more closely match the sampled wave. Furthermore, the segments are actually more fine-grained at the origin than at the high and low ranges. This is because most of the human speech we are trying to capture accurately is in this center range of the scale; there are fewer sounds at the very highest and lowest values.

© 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.

FIGURE 11 Quantizing the Digital Sample

Encoding the signal is a simple process. We have a single 8-bit code word to identify whether the analog signal was a positive or negative voltage, what value the signal was quantized to (which segment), and finally, which interval is represented by the code word. The first bit identifies either positive voltage (1) or negative (0). The next three bits represent the segment. There are eight segments in the positive range and eight segments on the negative range, so three bits provide the necessary encoding for the quantization. The last four bits identify the interval. A code word example is shown next: 10011100 In this case, the first 1 indicates a positive voltage; the next digits of 001 indicate this is the first segment (on the positive side), and 1100 indicates the twelfth interval. The code word is 8 bits; we generate a code word 8000 times per second (the sample rate). This gives us a bitrate output of 8 x 8000 = 64,000 bps (64 kbps). The process we just described is known as Pulse Code Modulation (PCM) and is the standard for uncompressed digital voice in telephony. One voice stream thus requires 64k of bandwidth for transport.

© 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.

NOTE The determination of voice quality is based on the Mean Opinion Score (MOS). This is a subjective measurement, created by gathering the opinion of live human listeners. A sample recording is played, and the listeners give it a score out of 5, where 5 is best. The same sample is played using different compression or processing methods and scored again. Because MOS is so subjective, other quality measurements exist that are more empirical and more accurate. For reference, standard PCM encoding (G.711) scores 4.1, and G.729 scores 3.92.

Compression is not a required step, but it is often done to save bandwidth in VoIP environments. The two main types of compression we are concerned with are the following: • Adaptive Differential P C M (ADPCM): This method does not send entire code words, but instead sends a smaller code that represents the difference between this word and the last one sent. This is not commonly used today, because it produces lower voice quality and compresses down only to about 16 kbps. • Conjugate Structure Algebraic Code Excited Linear Prediction (CS_ACELP): As the name suggests, this is more complex compression. Based on a dictionary or codebook of known sounds made by a standardized American male voice, the digital sample is analyzed and compared to the dictionary. The dictionary code that is the closest to the sample is sent. The codebook is constantly learning. The output of this compression is typically 8 kbps—with very little degradation of voice quality. This compression is widely used in VoIP.

Time Division Multiplexing (TDM)
T D M is the primary technology used in traditional digital voice; it is also extensively used in data circuits. The basic premise is to take pieces of multiple streams of digital data and interleave them on a single transmission medium.

T1 Circuits
On a Tl circuit, there are up to 24 channels available for voice. 64k from conversation 1 is loaded into the first Tl channel, then 64k from the conversation 2 is loaded into the second channel, and so on. If not enough conversations exist to fill the available channels, they are padded with null values. The 24 channels are grouped together as a frame. Depending on the implementation, either 12 frames are grouped together as a larger frame (called SuperFrame or SF), or

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These can be used to signal more advanced supervisory functions. and supervisory messages. C. Channel Associated Signaling (CAS)—T1 Although the 64 k channels of a Tl are intended to carry digitized voice. In ESF. Using RBS means that a slight degradation occurs in voice quality because every sixth frame has only 7 instead of 8 bits to represent the sample. SF implementation takes 12 frames and creates a SuperFrame. 24 channels are in an Extended SuperFrame. T l s are typically full duplex. with some El services becoming available in the United States. and 16 frames are grouped together as a multiframe. . with two wires sending and the other two wires receiving. however. of which 30 can be used for voice. which gives A.24 frames are grouped together (called Extended SuperFrame or ESF). this is not generally a perceptible degradation. addressing. the least significant bit of each channel in every sixth frame is "stolen" to generate signaling bit strings. Because CAS takes one bit from each channel in every sixth frame. (The other two are used for framing and signaling. respectively. In CAS circuits. This publication is protected by copyright. El circuits are common in Europe and Mexico. © 2008 Cisco Systems Inc. Using one bit per channel in every sixth frame gives two 12-bit signaling strings (known as A and B) per SuperFrame.) The 32 channels are grouped together as a frame. addressing. we must also be able to transmit signaling information. and so forth. it is known as Robbed Bit Signaling (RBS). Please see page 147 for more details. E1 Circuits An El is very similar to a T l . There are 32 channels. and D signaling strings. B. such as on-hook and off-hook. The A and B strings are used to signal basic status. All rights reserved.

. the signaling is outof-band in its own timeslot.Channel Associated Signaling (CAS)—T1 El signaling is slightly different. instead of generating A B C D bits. In an El CAS circuit. The 17th channel (channel 16 or timeslot 17) contains signaling information—no bits are robbed from the individual channels. This is the function of the D channel in an ISDN PRI or BRI implementation. © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. an ISDN BRI circuit provides two 64 k B channels and one D channel of 16 k.1 6 and 18-32 carry the voice data. a protocol known as Q. Common Channel Signaling (CCS) CCS provides for a completely out-of-band signaling channel. the first channel (channel 0 or timeslot 1) is reserved for framing information. Each channel has specific bits in timeslot 17 for signaling. Timeslots 2 . it is still considered C A S . An ISDN PRI Tl provides 23 voice channels of 64 k each (called Bearer or B channels) and one 64 k D (for Data) channel (timeslot 24) for signaling.931 is used out-of-band in a separate channel for signaling. This means that although El CAS does not use RBS. Please see page 147 for more details. An ISDN PRI El provides 30 B channels and 1 D channel (timeslot 17). however. The full 64 k of bandwidth per channel is available for voice.

and the factors that can cause problems in VoIP networks and how they can be mitigated. The digitized voice is then packaged in an appropriate protocol structure to move it through the IP infrastructure. and more importantly. and transport protocols. digitization. address and supervisory signaling. DSPs also change from one codec to another. A Layer 2 header of the correct format is applied. to change from a digital circuit to packetized voice. All rights reserved. © 2008 Cisco Systems Inc. DSPs DSPs are specialized chips that perform high-speed codec functions. Real-Time Transport Protocol (RTP) RTP was developed to better serve real-time traffic such as voice and video. timestamps. VoIP uses Digital Signal Processors (DSP) for the codec functions. This publication is protected by copyright. and sequence numbering. it now becomes just another binary payload to move around in a packet.U n d e r s t a n d i n g VoIP The elements of traditional telephony—status. for VoIP to interact with the PSTN properly. then by IP. Because we know how to digitize our voice. The D S P calculator on cisco. . then by UDP. Please see page 147 for more details. U D P provides multiplexing and checksum capability. DSPs are a vital component of a VoIP system. Different chip types have varying capacities.com will help you calculate what you must have. Understanding Packetization IP networks move data around in small pieces known as packets. the components of a VoIP network. but the general rule is that you want as many D S P resources available to you as possible. This section examines packetizing digital voice. DSPs are found in the IP phones to encode the analog speech of the user and to decode the digitized contents of the packets arriving from the other end of the call. RTP provides payload identification. DSPs are also used on IOS gateways at the interface to PSTN circuits. Voice payloads are encapsulated by RTP. or from an analog circuit to packetized voice. A single voice call generates two one-way RTP/UDP/IP packet streams. the type obviously depends on the link technology in use by each router interface. and so on—must have functional parallels in the VoIP world for systems to function as people expect them to. and other telephony features. signaling. allow conferencing and call park.

If we divide 8000 by 160. We still have 160 samples.729 Annex B variant: Allows the use of Voice Activity Detection and Comfort Noise Generation.711 (64kbps)—Toll-quality voice. lower audio quality than G. If we use compression. There is no provision for retransmission of a lost RTP packet.729 codec.Payload identification allows us to treat voice traffic differently from video. RTCP monitors the quality of the RTP stream. allowing devices to record events such as packet count. but reduced payload size. G. simply by looking for the RTP header label. per second. we can squeeze the 160-byte payload down to 20 bytes using the G. delay. for a one-way voice stream. still 20 msec of audio. Each RTP stream is accompanied by a Real-Time Transport Control Protocol (RTCP) stream. allows more voice channels encoded per DSP chip. The following tables summarize the additional overhead added by packetization and Layer 2 encapsulation (assume 50 packets per second (pps): • © 2008 Cisco Systems Inc. we see that we are generating 50 packets with 160 bytes of payload. Please see page 147 for more details. for example. . Codecs The codecs supported by Cisco include the following: • • G. Timestamping and sequence numbering allows VoIP devices to reorder RTP packets that arrived out of sequence and play them back with the same timing in which they were recorded. All rights reserved. eliminating delays or jerkiness. A single voice packet by default contains a payload of 20 msec of voice (either uncompressed or compressed). the actual bandwidth used by a single (one-way) voice stream can be significantly larger.729-A The values for bandwidth shown do not include the Layer 3 and Layer 2 overhead. This publication is protected by copyright. simplifying our configuration tasks. and jitter (delay variation). 20 msec gives us 160 samples. loss. uncompressed.729 or G. Because sampling is occurring at 8000 times per second. can be applied to G.729 (8kbps) • Annex A variant: less processor-intensive.

4 kbps) 18 Bytes 6 Bytes 6 Bytes 78 Bytes 31.000 bps (24 kbps) Bandwidth Calculation.400 (26. Layer 2 Total Bitrate incl.4 kbps) 66 Bytes 26.400 (82.4 kbps) When using G.) © 2008 Cisco Systems Inc.200 (31. (Note: Ethernet is not included because it is not classified as a slow link. . Without Layer 2 Codec G.000 bps (80 kbps) 20 Bytes 12 Bytes 8 Bytes 20 Bytes 60 Bytes 24.729. All rights reserved.400 (82.711 G. 12 Total incl.Bandwidth Calculation. The effect of using cRTP is illustrated in the following table.2 kbps) 206 Bytes 82.729 Voice Payload RTP Header UDP Header IP Header Total Before Layer 2 Total Bitrate @ 50 pps 160 Bytes 12 Bytes 8 Bytes 20 Bytes 200 Bytes 80. Please see page 147 for more details.711 = 2 0 0 Bytes/packet G.2 kbps) 66 Bytes 26. Layer 2 (@ 50 pps) 18 Bytes 6 Bytes 6 Bytes 218 Bytes 87. With Layer 2 Layer 2 Type G. This consumes significant bandwidth just for header transmission on a slow link.729 = 60 Bytes/packet Ethernet Multilink PPP Frame Relay FRF. cRTP reduces the RTP/UDP/IP header to 2 bytes without checksums or 4 bytes with checksums. the RTP/UDP/IP header of 40 bytes is twice the size of the 20B voice payload. The recommended solution is to use Compressed RTP (cRTP) on slow WAN links. This publication is protected by copyright.400 (26.200 (87.4 kbps) 206 Bytes 82.

All rights reserved. © 2008 Cisco Systems Inc. the VAD feature can be enabled.729 Voice Payload cRTP header w/ chksum cRTP header no chksum Total before Layer 2: Multilink PPP or Frame Relay FRF. VAD typically causes more problems than it solves. In situations where bandwidth is very scarce. causing the voice stream to be stopped during periods of silence. VAD should not be taken into account during the network design bandwidth allocation process because its effectiveness varies with background noise and speech patterns.Bandwidth Calculation. The theory here is that the bandwidth otherwise used for silence can be reclaimed for voice or data transmission. VAD also adds Comfort Noise Generation (CNG). Please see page 147 for more details. 12 Total WAN bandwidth @50 pps incl. by default silence is packetized and transmitted. Layer 2: 160 Bytes 4 Bytes 2 Bytes 164 Bytes 162 Bytes 20 Bytes 4 Bytes 2 Bytes 24 Bytes 22 Bytes 6 Bytes 68000 bps (68 kbps) 67. consuming the same bandwidth as speech.2 kbps) 6 Bytes 12. which fills in the dead silence created by the stopped voice flow with white noise.2 kbps) Voice Activity Detection (VAD) Phone conversations on average include about 3 5 % silence. VAD is also made ineffective by Music on Hold and fax features. and it is usually wiser to add the necessary bandwidth.200 bps (67.711 G. This publication is protected by copyright. . In Cisco Unified Communications. Using cRTP Codec G.200 (11. In reality.000 bps (12 kbps) 11.

• • Echo Cancellation: DSPs provide the calculation power needed to analyze the audio stream and filter out the repetitive patterns that indicate echo. This publication is protected by copyright. © 2008 Cisco Systems Inc. All rights reserved. echo cancellation is an important function. Please see page 147 for more details. perhaps for transit across a slow WAN link. MTPs provide a point for the stream to be terminated while other services are set up. Echo is a chief cause of perceived poor voice quality. D S P resources are used for the following: • Conferencing: DSPs mix the audio streams from the conference participants and transmit the mix (minus their own) to each participant. Transcoding and Media Termination Points (MTP): A transcoder changes a packetized audio stream from one codec to another. .Additional DSP Functions In addition to digitizing voice.

. and teardown functions of VoIP calls.323. it is an umbrella standard that defines several other related protocols for specific tasks. Protocol Standard? Inter-Vendor Compatibility Implemented on Gateways Implemented on Cisco IP Phones Operating Mode H.323 is not itself a protocol. The signaling protocol in use must pass the supervisory. Originally conceived as a multimedia signaling protocol to emulate traditional telephony functionality in IP L A N © 2008 Cisco Systems Inc. and address information expected in any telephony system.323 H. the endpoints have the intelligence to perform the call-control signaling themselves. The following table summarizes the characteristics of the four signaling protocols dealt with here. also third-party phones Cisco IP Phones only Peer-to-Peer Client/Server Peer-to-Peer Client/Server H. VoIP signaling protocols are either peer-to-peer or client-server. including SCCP. in the case of peer-to-peer protocols. This publication is protected by copyright. MGCP. A number of different protocols are in use—some standards-based. VoIP Signaling Protocols VoIP signaling protocols handle the call setup.323 MGCP SIP SCCP Yes--ITU Yes--IETF Yes--IETF N o . The following sections introduce the signaling protocols you should know about.-Cisco Proprietary Very Good Good Basic Cisco only Yes Yes Yes Some No No Yes. All rights reserved.I n t r o d u c i n g VoIP Signaling Protocols VoIP signaling protocols are responsible for call setup. It is important to keep in mind that the signaling functions are an entirely separate packet stream from the actual voice bearer path (RTP). maintenance. and teardown. maintenance. H. and each with advantages and disadvantages. and SIP. Client-server protocols send event notifications to the call agent (the Unified CM server) and receive instructions on what actions to perform in response. Please see page 147 for more details. informational. others proprietary.

SCCP uses TCP connections to the Unified CM to set up. and the call agent instructs the gateway on what to do. and because it is text-based. SIP Session Initiation Protocol is an IETF standard that uses peer-to-peer signaling. and tear down voice and video calls. Please see page 147 for more details. It is simple to configure and allows the call agent to control the M G C P gateway.environments. All rights reserved. It is very similar in structure and syntax to HTTP. and the 500 Series) and some gateways support SCCP.323 is supported by all Cisco voice gateways and CM platforms as well as some third-party video endpoints. The gateway reports events such as a trunk going off-hook. although many also support SIP. One of its most important capabilities is creating SIP trunks to IP Telephony service providers. SIP does not yet support the full feature set available to SCCP phones. SIP can use multiple transport layer protocols and can support security and proxy functions. it is a long-established and stable protocol very suitable for intervendor compatibility. the gateway has no local dial plan because all call routing decisions are made at the call agent and relayed to the M G C P gateway. maintain. . M G C P is not as widely implemented as SIP or H. It is referred to as a stimulus protocol. MGCP Media Gateway Control Protocol is a lightweight client/server protocol for PSTN gateways and some clients. M G C P is not supported by Unified CM Express or the Smart Business Communication System. further developments and extensions to the standard will soon make it feature-comparable with SCCP.323. CM Express. SCCP is the default signaling protocol for all Cisco IP phones. This publication is protected by copyright. H. meaning that it sends messages in response to events such as a phone going off-hook or a digit being dialed. replacing or enhancing traditional T D M PSTN connections. eliminating the need for expensive gateways with intelligence and complex configurations. SCCP Skinny Client Control Protocol is a Cisco-proprietary signaling protocol used in a client-server manner between Unified CM and Cisco IP Phones (and some Cisco gateways). it is relatively simple to debug and troubleshoot. SIP is an evolving standard that currently provides basic telephony functionality. © 2008 Cisco Systems Inc. All Cisco Unified Communications call agents (CM.

it provides the physical connection and logical translation between two or more different network technologies. The device that acts as the interface to the PSTN is the voice gateway. and to do so we need to connect to a phone service provider. Cisco gateways support two types of dial peers: POTS and VoIP. it is associated with an inbound port. There will be inbound and outbound call legs at each gateway router. . the dial peers may be POTS inbound and VoIP outbound. and the outbound call leg is routed to a destination dial peer. (This is the inbound call leg. Gateways In the Cisco Unified Communications architecture. Call Legs A call leg is the inbound or outbound call path as it passes through the gateway. and E & M or digital T l / E l or PRI interfaces. or possibly both VoIP. All rights reserved. Dial peers identify the source and destination endpoints of call legs. and Dial Peers The following sections establish some terms of reference. a gateway is typically a voice-enabled router with an appropriate voice port installed and configured. F X S . It is unlikely but not impossible that the © 2008 Cisco Systems Inc. Gateways can have both analog and digital voice port connections. Dial Peers A dial peer is a pointer to an endpoint. POTS dial peers are addressed with PSTN phone numbers. including analog FXO.) As the call is sent out another gateway port. Depending on the direction of the call. and VoIP dial peers are addressed by IP addresses. Please see page 147 for more details. This publication is protected by copyright. vice versa.C o n n e c t i n g a VoIP S y s t e m to a S e r v i c e Provider Network A VoIP system that can place calls only to other devices on the IP network is only marginally useful. whether via traditional T D M links or ITSP connections. an inbound call leg is matched to a dial peer. Understanding Gateways. this creates the outbound call leg. Voice Ports. we still need to place calls out to the PSTN. As the call comes into the gateway. identified by an address (a pattern of digits).

Each dial peer also defines attributes such as the codec to use. The partial output that follows shows a simple POTS dial peer configuration: G a t e w a y ( c o n f i g ) # d i a l . although dial peer 0 exists by default and cannot be deleted. the destination pattern could men be "." would match all strings from 200 through 209. * and # . The specified pattern can be a specific phone number (as above. The destination-pattern command associates a phone number with a dial peer. "20[4-6]" defines 204. long-distance.T". this is useful in cases where local. and international PSTN numbers may be called. 8675309) or an expression that defines a range of numbers. *. The following table briefly explains destination-pattern syntax. these are valid DTMF digits. The router uses the patterns to decide which dial peer (and associated physical port) it should route a call to. and other feature settings.p e e r v o i c e 10 p o t s Gateway(config-dialpeer)#destination-pattern Gateway(config-dialpeer)#port 1/0/1 8675309 The number assigned to dial peers is arbitrary. The pattern "20. Specifies any one wildcard digit (0 . This publication is protected by copyright. All rights reserved. This pattern would match any string of up to 32 digits. using either the CLI or GUI interface. Please see page 147 for more details. The keyword pots creates a POTS dial peer. Square brackets define a range. within which any one digit may be matched. . and 206. Character + Meaning The preceding digit is repeated one or more times.9. The port command identifies the physical hardware connection the dial peer will use to reach the destination pattern. plus 20*and20#. the keyword voip would create a VoIP dial peer. NOT wildcards. Dial peers are configured in the gateway IOS. (dot) [] T © 2008 Cisco Systems Inc. for example.inbound and outbound dial peers would both be POTS. (comma) . QoS settings. #). Indicates a variable length string. The destination-pattern command identifies that the attached device (phone or PBX) terminates calls to the specified number (or a range of numbers if connecting to a PBX). Inserts a one-second pause. 205.

. it should be a loopback IP so that the address is always available even if a physical interface fails.. Look for the answer-address command in a dial peer that matches the calling number or ANI string of the inbound call leg. at which point the call is routed to the outbound dial peer configured with that matching pattern. Look for the POTS dial peer port command that matches the voice port of the incoming call (POTS dial peers only). Routers attempt to match dial peers for the inbound call leg according to the following rules: NOTE The default dial peer 0 cannot be deleted or modified. 3. the destination pattern is any four-digit number starting with "4. Please see page 147 for more details. It does not negotiate services such as VAD. Each successive digit may validate some patterns while eliminating others until a single pattern represents the longest match between the dialed digits and the destination pattern. If the IP address is on a router. 2. DTMF Relay. The dial peer 0 configuration for inbound VoIP calls contains the following: • • Any codec VAD enabled 1.1. 5. This publication is protected by copyright. Look for the destination-pattern command in a dial peer that matches the calling number or ANI string of the inbound call leg. . it attempts to find the longest match.323 dial peer (in contrast to a SIP dial-peer). is used to identify the IP (version 4 in this case) address of the gateway or call agent that will terminate the call. session-target. ipv4:10. If all of the above fail to match. match against Default Dial Peer 0 as a last resort. Look for the incoming called-number command in a dial peer that matches the called number or DNIS string of the inbound leg.2 In this example. 4. • No RSVP Support • Fax-rate voice © 2008 Cisco Systems Inc." A new command.1.p e e r voice 20 voip Gateway(config-dialpeer)# Gateway(config-dialpeer)# destination-pattern session target 4.. The default dial peer 0 config for inbound POTS calls includes the following: • no ivr application When a router is matching the dialed digits against the patterns in the configured dial peers. The preceding configuration creates an H. All rights reserved.C o n n e c t i n g a VoIP S y s t e m to a S e r v i c e P r o v i d e r N e t w o r k Configuring VoIP dial peers is equally simple. or TCL applications. This occurs on a digit-by-digit basis. Examine the following configuration: G a t e w a y ( c o n f i g ) # d i a l .

Consider the following configuration:
d i a l - p e e r voice 10 voip destination-pattern session t a r g e t i d i a l peer v o i c e 2 0 v o i p destination-pattern 867[2-3]... session t a r g e t ipv4:10.10.20.1 .T ipv4:10.10.10.1

!
d i a l - p e e r voice 30 voip destination-pattern 8674... session t a r g e t i p v 4 : i d i a l - p e e r voice 40 voip d e s t i n a t i o n - p a t t e r n 8675309 session t a r g e t ipv4:10.10.40.1 10.10.30.1

Given this configuration, the following example dialed numbers illustrate how the patterns match dialed digits: • • • • The dialed number 867-5309 will match dial peer 40 (exact 7-digit match) The dialed number 867-4309 will match dial peer 30 (first four digits match) The dialed number 867-3309 will match dial peer 20 (first four digits match) The dialed number 876-5309 will match dial peer 10 (no other exact match, so the " . T " pattern matches)

Internet Telephony Service Providers
As VoIP technology matured and stabilized, telephone service providers began extending VoIP connectivity to their customers, allowing for simple, flexible connection alternatives to traditional T D M links. Internet Telephony Service
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Providers (ITSP) connections are typically much less expensive, available in smaller bandwidth increments than Tl or PRI links, and can route nonvoice data traffic concurrently. QoS configuration is supported (and in fact is required for proper VoIP operation). Most ITSP links use SIP, but H.323 is an option. The gateway configuration is relatively simple, with the creation of a VoIP dial peer pointing at the provider with the parameters they supply. PSTN calls are routed to the provider, who then routes calls to their PSTN connection, usually with a toll-minimizing route that dramatically reduces long-distance costs to the customer.

Understanding Call Setup and Digit Manipulation
Successfully completing a phone call requires that the correct digits are sent to the terminating device, whether on the VoIP network or the PSTN. PSTN calls are typically more complex because of the varying local and international requirements for the number of digits required to route the call. Over and above this basic requirement are the additional complexities imposed by requirements of the business: we may want to change our A N I number, add or strip access codes, compensate for undesirable default behavior, or build specialized functionality for our particular purposes. This section deals with digit manipulation and troubleshooting.

Digit Consumption and Forwarding
Some strange things happen when an IOS gateway matches a dial peer for an outbound call leg and forwards the dialed digits to the terminating device. For POTS dial peers, the gateway consumes (meaning strips away) the left-justified digits that exactly match the dial-peer destination pattern and forwards only the wildcard-matched digits to the terminating device. Clearly, this could cause problems if the PSTN were to receive only 4 digits, as in this example:
d i a l - p e e r voice 20 pots destination-pattern port 1/0:1 867....

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With this configuration, if the dialed number was 867-5309, the gateway would forward only 5309 (the wildcard matches), and the PSTN would be unable to route the call. Adding the command no digit-strip in the dial-peer configuration will change this behavior and cause the gateway to forward all dialed digits. For VoIP dial peers, the default behavior is to forward all collected digits.

Digit Collection
The router will collect digits one at a time and attempt to match a destination pattern. As soon as it has an exact match, the call is immediately placed, and no more digits are collected. If there are destination patterns that have overlapping digits, this can cause calls to be misrouted, as in the following example:
Dial-peer voice 1 voip D e s t i n a t i o n p a t t e r n 555 Session t a r g e t ipv4:10.1.1.1

!
Dial-peer voice 2 voip D e s t i n a t i o n - p a t t e r n 5552112 Session t a r g e t ipv4:10.2.2.2

If the user dials 555-2112, dial peer 1 will exactly match at the third digit, the call will be immediately routed using dial peer 1, and only the collected digits of 555 will be forwarded. We solve the problem by changing the configuration as shown next:
Dial-peer voice 1 voip Destination pattern 5 5 5 . . . . Session t a r g e t ipv4:10.1.1.1

!
Dial-peer voice 2 voip D e s t i n a t i o n - p a t t e r n 5552112 Session t a r g e t ipv4:10.2.2.2

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. Adding the command prefix 6045552 forces the router to prepend the additional digits required to route the call over the PSTN: d i a l . change. We do this to avoid inconveniencing users or to match the dialed digit requirements of a gateway or the PSTN. This publication is protected by copyright. Dial peer 1 is also a match. All rights reserved. prefix The prefix dial-peer command adds digits to the beginning of the string after the outbound dial peer is matched but before passing digits to the destination.. forward-digits forward-digits: This dial-peer command forces the specified number of digits to be forwarded. the router determines that dial peer 2 is an exact match and immediately places the call. or remove digits before the call is placed. You can specify a number of digits to forward (as shown in the example that follows) or use forward-digits all to force all dialed digits to be forwarded. overriding the default behavior of stripping the exact matches. when the third digit is entered. but because of the wildcards..Now. An example of its use is a POTS dial peer with 2 . d i a l . © 2008 Cisco Systems Inc. the router cannot make an exact match because both dial peers are possible matches. If the user dials 2112.000 possible numbers (0000 through 9999). . when the last digit is dialed. the default behavior is for the POTS dial peer to forward only 112. . it is not as close a match as dial peer 2. whether the digits were exact match or wildcard matches... as the destination pattern.p e e r voice 20 pots destination-pattern prefix port 6045552 1/0/0 2. as described in the following sections. the destination pattern matches 10. Please see page 147 for more details. We have several methods of modifying the digit string. Digit Manipulation Sometimes we need to add.p e e r voice 20 pots destination-pattern 5552.

5552. This command is applied before the outbound dial peer is matched... . . The match and replace patterns are identified and separated by the "/" characters that begin and end the patterns. Please see page 147 for more details. All rights reserved.r u l e 1 rule 1 /555/ /867/ The rule command defines a pattern to match (in this case 555) and a pattern to change the matched digits to (in this case 867). num-exp 2 . . DNIS. Define the translation rule globally: voice t r a n s l a t i o n . d i a l . © 2008 Cisco Systems Inc. This publication is protected by copyright. or redirect number for a call.p e e r voice 20 pots destination-pattern port 1/0/0 Translation Rules voice translation-rule: This global command configures number translation profiles to allow us to alter the ANI. 5552.forward-digits 7 port 1/0/0 Number Expansion num-exp: The number expansion table is a global command that either expands an extension (perhaps a 4-digit extension into a full 10-digit PSTN number) or completely replaces one number with another... so there must be a configured dial peer that matches the expanded number for the call to be forwarded. Using the command is a three-step process: 1.

Apply the profile to one or more dial peers. Cisco Regular Expression Characters for Voice Translation Rules Character Description Matches any single character. continues © 2008 Cisco Systems Inc.p e e r voice 20 pots description translated destination-pattern port 1/0/0 t o Jenny t r a n s l a t i o n . Identifies the start and end of both the match and replace phrases. Do not match a single character specified in the list. This publication is protected by copyright..p r o f i l e JENNY translate called 1 3. which can be quite complex. Match the expression at the beginning of the digit string. Match a single character in a list. All rights reserved. In this example we are translating the called number: v o i c e t r a n s l a t i o n . Translation rules use regular expression syntax. In the replace phrase: Reference a set number from the match phrase. \ (match) \(replace) A In the match phrase: Escape the special meaning of the next character. Create the voice translation profile containing the translate instruction (the options are [calledlcallingl redirect-calledlredirect-target].2.p r o f i l e outgoing JENNY 5552. The following table defines the characters used. $ / [0-9] [ 0-9] A . Please see page 147 for more details. and examples follow. Match the expression at the end of the digit string. either inbound or outbound: d i a l . and reference the rule we just defined by number..

Cisco Regular Expression Characters for Voice Translation Rules *
+

continued

Repeat the previous expression 0 or more times. Repeat the previous expression 1 or more times. Repeat the previous expression 0 or 1 time. Identifies a set in the match expression.

?
() Example 1:
rule 1

/123/ /456/

The first set of forward slashes defines the match phrase; the second set defines the replace phrase. This expression means "match 123 and replace it with 456." Thus: • • • • 123 is replaced with 456 6123 is replaced with 6456 1234 is replaced with 4564 1234123 is replaced with 4564123 (only the first instance of the match is replaced)

Example 2:
voice t r a n s l a t i o n ? r u l e 1 rule 1 / 4 0 . . . / /66660B0/
A

This example replaces any five-digit number that begins with " 4 0 " with the number "6666000". Example 3:
voice t r a n s l a t i o n ? r u l e 1 / \(867\)\(....\)/
A

/555\2/

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This example means: "If the number starts with 867 and is followed by any four other digits, change the 867 to 555 and replace the other four digits with the digits in Set 2 of the match." Remember that the forward slashes define the match and replace phrases; the backslashes mean "the next character is not part of what to match"; the round brackets indicate which sets of characters in the matched number to keep in the replaced number. The sets are numbered starting with 1, so the first set of round brackets is 1, and the second is 2 (as in this example).

Private Line Automatic Ringdown (PLAR)
PLAR creates a permanent association between a voice port and a destination number (or voice port). When PLAR is configured, going off-hook on that voice port automatically dials the pattern specified by the connection plar <number> command. The caller does not hear a dial tone and does not have to dial a number. Think of PLAR as a hotline; pick up the Batline and you get Batman without having to dial. The following shows a simple P L A R configuration that will call 867-5309 when the phone goes off-hook:
voice port 1/0/0 p l a r 8675309

connection

Troubleshooting Dial Plans and Dial Peers
The following sections discuss some of the commands available to troubleshoot your configuration.

show dial-peer voice
To display information for voice dial peers, use the show dial-peer voice command in user EXEC or privileged EXEC mode.
show d i a l - p e e r v o i c e [number | summary]

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Syntax Description number
summary

(Optional) A specific voice dial peer. Output displays detailed information about that dial peer. (Optional) Output displays a short summary of each voice dial peer.

If both the name argument and summary keyword are omitted, output displays detailed information about all voice dial peers. The following is sample output from this command for a VoIP dial peer:
Router# show d i a l - p e e r v o i c e 101 VoiceOverIpPeer101 peer t y p e = v o i c e , description tag = 6001, = '', "6001', information type = voice,

destination-pattern = '', preference=0,
1

answer-address =

CLID R e s t r i c t i o n = None CLID Network Number = " CLID Second Number s e n t CLID O v e r r i d e RDNIS = d i s a b l e d , source c a r r i e r - i d = target c a r r i e r - i d = '', target trunk-group-label = ' ' , source t r u n k - g r o u p - l a b e l = numbering Type = "unknown' group = 6 0 0 1 , Admin s t a t e i s u p , O p e r a t i o n s t a t e i s up, i n c o m i n g c a l l e d - number = DTMF Relay = d i s a b l e d , <output omitted> connections/maximum = 0 / u n l i m i t e d ,

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All rights reserved. V o i c e . Please see page 147 for more details. C a l l i n g Number=4085550111.p e e r hunt 0 PASS TAG TYPE 100 p o t s 101 v o i p 102 v o i p 99 v o i p 33 p o t s ADMIN OPER PREFIX up up up up up up up up down down 5550112 5550134 DEST-PATTERN PREF THRU SESS TARGET 0 0 0 0 0 syst ipv4 10. This publication is protected by copyright.1 1 syst ipv4 10.735: Result=Success(0) C a l l i n g Number=. . Peer Encap Type=ENCAP_VOIP. use the debug voip dialpeer command in privileged EXEC mode.p e e r = 1 0 0 //-1/6372E2598012/DPM/dpMatchPeersCore: Peer I n f o Type=DIALPEER_INFO_SPEECH a f t e r DP_MATCH_INCOMING_DNIS.10. Peer Search Type=PEER_TYPE_VOICE. C a l l e d Number=3600.731: //-1/6372E2598012/DPM/dpAssociateIncomingPeerCore: Incoming D i a l .p e e r v o i c e summary d i a l .l n t e r f a c e = 0 x 0 . Peer *May *May I n f o Type=DIALPEER_INFO_SPEECH //-1/6372E2598012/DPM/dpAssociateIncomingPeerCore: Incoming D i a l .p e e r = 1 0 0 //-1/6372E2598012/DPM/dpAssociateIncomingPeerCore: Result=Success(0) a f t e r DP_MATCH_INCOMING_DNIS. Router# debug v o i p d i a l p e e r i n o u t v o i p d i a l p e e r i n o u t debugging i s o n *May *May 1 19:32:11.1 1 syst debug voip dialpeer inout To display information about the voice dial peers.731: 1 19:32:11. Timeout=TRUE.731: 1 19:32:11.The following is sample output from this command with the summary keyword: Router# show d i a l . C a l l e d Number=3600. © 2008 Cisco Systems Inc. 1 19:32:11.10.

L i n e Code i s AMI. 0 Fr Loss Sees. 0 Path Code V i o l a t i o n s 0 S l i p Sees. © 2008 Cisco Systems Inc. Please see page 147 for more details. This publication is protected by copyright. Applique type is Channelized T1 Cablelength is short No a l a r m s d e t e c t e d . All rights reserved.735: 19:32:11. The following is sample output from this command: Router# show c o n t r o l l e r s T1 4 / 1 i s u p . Clock Source i s l i n e Data i n c u r r e n t i n t e r v a l (10 seconds e l a p s e d ) : 133 t1 0 L i n e Code V i o l a t i o n s .*May *May *May 1 1 1 19:32:11.735: 19:32:11. Result=Success(0) Result=SUCCESS(0) The following event shows the matched dial peers in the order of priority: List of Matched Outgoing Dial Peer(s): 1: Dial PeerTag=3600 2: Dial Peer Tag=36 Troubleshooting Signaling for POTS Call Legs show controllers t1 The show controllers tl command displays Tl (or E l ) controller status and function. Framing i s ESF.735: //-1/6372E2598012/DPM/dpMatchPeersCore: C a l l e d Number=3600 //-1/6372E2598012/DPM/dpMatchPeersCore: a f t e r DP_MATCH_DEST //-1/6372E2598012/DPM/dpMatchPeersMoreArg: Match Rule=DP_MATCH_DEST. .

show voice port Use the show voice port command to display configuration and voice-interface-card-specific information about a specific port. port 1/0/0 Port is 0 Sub-unit is 0. The following is sample output for an E & M analog voice port: Router# show v o i c e E&M S l o t i s 1 . no alarms were detected. Receiver is getting AIS. Possible alarms are as follows: • • • • • • • Transmitter is sending remote alarm. 0 S e v e r e l y E r r Sees. Transmitter is sending AIS. All rights reserved. Receiver has loss of signal. 0 Degraded Mins 0 E r r o r e d Sees. Please see page 147 for more details. This publication is protected by copyright.C o n n e c t i n g a VoIP S y s t e m to a S e r v i c e P r o v i d e r N e t w o r k 0 L i n e E r r Sees. 0 B u r s t y E r r Sees. 0 U n a v a i l Sees In this output. Type of V o i c e P o r t is E&M O p e r a t i o n S t a t e is DORMANT A d m i n i s t r a t i v e S t a t e is UP I n i t i a l Time Out i s s e t t o 0 s I n t e r d i g i t Time Out i s s e t t o 0 s © 2008 Cisco Systems Inc. Receiver has remote alarm. . Receiver has loss of frame. Receiver has no alarms.

''. © 2008 Cisco Systems Inc. : 1001 VoiceEncapPeer1003 i n f o r m a t i o n type = v o i c e . incoming called-number = DTMF Relay = d i s a b l e d . unknown' group = 1003. use the show dialplan number command in privileged EXEC mode. connections/maximum = 0 / u n l i m i t e d . Router# show d i a l p l a n number 1001 Macro E x p . All rights reserved. huntstop = enabled. This publication is protected by copyright. answer-address = numbering Type = preference=0. Please see page 147 for more details.w i r e E&M Type is 1 D i a l Type i s dtmf In Seizure Out S e i z u r e is inactive is i n a c t i v e show dialplan number To display which outgoing dial peer is reached when a particular telephone number is dialed. destination-pattern = ''.Analog Info Follows: Region Tone i s s e t f o r n o r t h a m e r i c a Voice card s p e c i f i c S i g n a l Type Info Follows: is wink-start O p e r a t i o n Type i s 2 . O p e r a t i o n s t a t e i s up. Admin s t a t e i s u p . prefix = ' ' . . 1 1001'. type = pots. t a g = 1003.

Passed. Id=-1 *Apr 18 2 0 : 4 2 : 1 9 . C a l l i n g Number=4085550111(TON=National. S c r e e n i n g = U s e r . Passed. *Apr 18 2 0 : 4 2 : 1 9 . NPI=Unknown). which serves as the interface between the call session application and the underlying network-specific software. Called Number=83103(TON=Unknown. you can see the call setup and teardown operations performed on both the telephony and network call legs. NPI=ISDN. //-1/9C5A9CA88009/CCAPI/ccCheckClipClir: NPI=ISDN. Source T r k g r p Route L a b e l = .forward-digits default 1 session-target = ' . Presentation=Allowed). Router# debug v o i p c c a p i i n o u t v o i p c c a p i i n o u t debugging i s o n NOTE This debug generates a very long output. //-1/9C5A9CA88009/CCAPI/cc_api_call_setup_ind_common: Call Info( T h e following lines show information about the calling and called numbers. You can use the output from this c o m m a n d to understand how calls are being handled by the router. Please see page 147 for more details. P r o g r e s s I n d i c a t i o n = N U L L ( 0 ) . Calling Number=4085550111(TON=National. This c o m m a n d shows how a call flows through the system. CLID T r a n s p a r e n t = F A L S E ) . 3 4 7 : Interface=0x64F26F10. This publication is protected by copyright. S u b s r i b e r Type S t r = R e g u l a r L i n e . I suggest you familiarize yourself with sample outputs from the Cisco IOS Debug Command Guide or better yet from your own lab experimentation. T h e network presentation indicator (NPI) shows the type of transmission. which is impractical to fully duplicate here. C a l l i n g Translated=FALSE. //-1/9C5A9CA88009/CCAPI/ccCheckClipClir: NPI=ISDN. The Incoming Dial-Peer field shows that the incoming dial peer has been matched. C a l l i n g I E Present=TRUE. Passed. Incoming Dial-peer=1. T a r g e t T r k g r p Route L a b e l = . All rights reserved. Using this debug level. 3 4 7 : Out: Presentation=Allowed) © 2008 Cisco Systems Inc. I n : C a l l i n g Number=4085550111(TON=National. v o i c e . S c r e e n i n g = U s e r .p o r t = '1/1' debug voip ccapi inout The debug voip ccapi inout c o m m a n d traces the execution path through the call control API. Screening=User. FinalDestinationFlag=TRUE. C a l l . 3 4 7 : Presentation=Allowed) *Apr 18 2 0 : 4 2 : 1 9 .

On many Cisco switches. This publication is protected by copyright. The switch is capable of sending the V V L A N ID using CDP messages. it can be an access port instead. (Each V L A N is a new. The port that connects to the network switch can act as an 802. A voice V L A N (VVLAN. the port connecting the phone does not need to be a trunk. . also called an Auxiliary VLAN) is an additional V L A N for the exclusive use of VoIP and video traffic. © 2008 Cisco Systems Inc. allowing both voice and data traffic to be multiplexed in their respective VLANs on the single cable to the network switch. Topics covered include the following: • • • • • Voice VLANs DHCP NTP Power over Ethernet IP Phone firmware and configuration files Voice VLANs VLANs provide a logical separation of Layer 3 traffic and are created at Layer 2 (the network switch). lq trunk. and the third port is an internal one for the voice traffic generated and received by the phone. Please see page 147 for more details. and the phone then sends frames from itself tagged with the learned V V L A N ID and forwards frames from the attached PC untagged. All rights reserved. the best practices components for preparing the network to properly support Unified Communications are explored. These untagged frames will be tagged with the access V L A N ID configured on the switch port when they are processed by the switch. a measure of additional security.) Most Cisco IP Phones are actually 3-port switches. The second port connects the desktop PC to the phone (and thus to the network over the trunk on the first port). and simpler deployment because you do not have to renumber the IP address scheme of the whole network to add VoIP endpoints.Preparing the Infrastructure to Support Unified Communications In this section. The benefits of using a V V L A N include isolation from the broadcast traffic data VLANs. separate subnet.

The following configuration is a typical example of router-based D H C P to support IP Phones: service dhcp ! e n a b l e s t h e DHCP s e r v i c e I i p dhcp e x c l u d e d .s e n s i t i v e name) and e n t e r s DHCP c o n f i g u r a t i o n mode I network 1 0 . Cisco routers have D H C P server capability. 2 5 5 . This publication is protected by copyright. using the 802. 1 . This can be done on an existing D H C P server.1. This is the recommended setting. .1.The phone adds a QoS marking to its own frames. 1 . 1 .10 s p e c i f i e s a s t a r t / e n d range of addresses t h a t DHCP w i l l NOT a s s i g n ip dhcp p o o l name IP_PH0NES ! C r e a t e s a p o o l of a d d r e s s e s ( c a s e . 2 5 5 . All rights reserved.1.a d d r e s s 1 0 . lq frame header Class of Service (CoS) field.1. 1 ! 10. or a new one can be added if necessary. Create a separate subnet for the Voice V L A N and add the Option 150 parameter to identify the T F T P server IP address. but it can be modified. 1 . The following is a typical switchport configuration for an attached IP Phone in V V L A N 100 and the PC in V L A N 50: Switch(config)#interface FastEthernet 0/1 S w i t c h ( c o n f i g .1 D e f i n e s t h e d e f a u l t gateway © 2008 Cisco Systems Inc.i f ) # s w i t c h p o r t mode access S w i t c h ( c o n f i g . 0 2 5 5 . The phone marks its frames as CoS 5 by default. 0 ! D e f i n e s t h e subnet o f a d d r e s s e s f o r t h e p o o l default-router ! address 10. Please see page 147 for more details.i f ) # s w i t c h p o r t access v l a n 5 0 Switch(config-if)#switchport voice vlan portfast 100 Switch(config-if)#spanning-tree DHCP It is recommended that you use D H C P for IP Phone addressing.

and for good voice performance.10 192. CM Express.3 I d e n t i f i e s t h e NTP master c l o c k address Sunday October 02:00 © 2008 Cisco Systems Inc. or SBCS).2.1.168. The call agent(s) are configured to get their time from a master clock. Network Time Protocol (NTP) is used on all Cisco devices to sync the system clock to a master clock. Network Time Protocol Clock synchronization is important in VoIP systems for accurate Call Detail Records (used for billing). Please see page 147 for more details. CM Business Edition. you will need to add the ip helper-address <ip-address> command on the Voice V L A N interface of the router so that it will forward D H C P broadcasts from the phones to the D H C P server. This publication is protected by copyright.s e r v e r address ! ! option ! 150 i p 192. R o u t e r ( c o n f i g ) # c l o c k timezone pst -8 ! S p e c i f i e s t h e l o c a l t i m e z o n e as PST (8 hours b e h i n d GMT) R o u t e r ( c o n f i g ) # c l o c k summer-time zone r e c u r r i n g f i r s t Sunday a p r i l 0 2 : 0 0 l a s t ! A c t i v a t e s Summer Time change in t h e s p e c i f i e d d a t e range ! Router(config)#ntp ! server 10.168.1.11 . easier troubleshooting and debugging. IP Phones get their time from the call agent (CM.2 i d e n t i f i e s t h e TFTP s e r v e r IP If you choose to use a D H C P server that resides on a different network.up to 8 I P ' s i d e n t i f i e s t h e DNS s e r v e r IP a d d r e s s ( e s ) 192.1. All rights reserved. .d n s .1.168. usually a highly accurate atomic or radio clock external to the network.

port. The IP Phone and the switch have a common PoE delivery method. directory URL. This publication is protected by copyright. © 2008 Cisco Systems Inc.3af standard Extra care should be taken to ensure the following: • • • RJ-45 cabling is tested and meets the required standard.Cisco IP Phone Firmware and XML Configuration Files Cisco IP Phones need the following three separate files to function: • The firmware file: This file is loaded into nonvolatile memory and is persistent across reboots. This file is created when the IP Phone has been added to the configuration. These files are downloaded by the phone during its boot process. The PoE switch has a suitable UPS backup to provide power continuance in the event of a power failure. There are two methods of PoE delivery: • • Cisco prestandard (inline power) 802. SEPAAAABBBBCCCC.xml: This is the X M L configuration file that devices use if their specific SEP<MAC> file is not available (typically if they have not registered before or if they have been factory reset). Power over Ethernet Power over Ethernet (PoE) is a desirable option because it eliminates the cost and clutter of power bricks for the IP Phones. All rights reserved. To make the firmware files available to the phones. • • XMLDefault. locale.cnf. and many other pieces of information.cnf. use the router command tftp-server flash:firmware-file-name. which specifies the IP address. The command load phone-type firmware-file is also required to associate the model of IP phone with the appropriate firmware file.xml: This is the device-specific X M L configuration file ( A A A A B B B B C C C C is the MAC address of the phone). . Please see page 147 for more details. firmware.

The switch sends a special tone. Please see page 147 for more details. . 5. or a variety of power injectors are available. © 2008 Cisco Systems Inc. 2. When the switch receives the returning FLP. The phone uses C D P to specify its power needs.3af PoE negotiation steps: 1. This publication is protected by copyright. The F L P goes to the powered device. The Cisco prestandard PoE method works as follows: 1. resulting in the FLP arriving back at the switch. 4. 2. it applies power to the line. The powered device (IP phone) boots.3af-compliant device will apply 25 ohms resistance across the DC circuit. 4. called a Fast Link Pulse (FLP). The switch detects the resistance and applies low-power PoE to the link. 3. the switch will therefore never receive the FLP from a device that does not require PoE. The standard requires that all eight pins in the RJ-45 cable be present and punched down. The switch applies constant DC power to all ports that may require PoE. The following describes the 802. 5. An 802. the PoE device has a physical link between the pin on which the F L P arrives and a pin that goes back to the switch. The powered device (the phone) boots.) The 802. in this case an IP phone. (Power requirements vary from device to device.Alternatively. Using CDP. All rights reserved.3af PoE standard works slightly differently. Non-PoE devices will not have this link. When unpowered. out of the port. 3. This creates a circuit. The link comes up within 5 seconds. the IP Phone tells the switch exactly how much power it needs. 6. each IP Phone may be powered by its own cable and transformer.

They do so not by creating additional bandwidth. this often means that data applications and protocols are restricted from accessing bandwidth when VoIP traffic needs it. but by controlling how the available bandwidth is used by the different applications and protocols on the network. This section defines and describes why QoS is needed and explains how to configure and deploy a QoS solution using both the Modular QoS Command Line (MQC) and AutoQoS. or unacceptably long pauses in the conversation that cause overlap. or one talker interrupting the other. however. QoS configurations provide bandwidth guarantees while minimizing delay and jitter for priority traffic like VoIP. This publication is protected by copyright. In effect. This does not have much of an impact on the data traffic. All rights reserved. The areas that QoS can address to improve and guarantee voice quality are the following: • • • Bandwidth Delay (including delay variation or jitter) Packet loss © 2008 Cisco Systems Inc. . because it is generally not as delay or drop-sensitive as VoIP traffic. echo. QoS Definition QoS is defined as The ability of the network to provide better or "special" service to a set of users and applications at the expense of other users and applications.Q u a l i t y of S e r v i c e Quality of service (QoS) is possibly the single most important feature to deploy to ensure a successful VoIP system. Voice and video traffic is very sensitive to delayed packets. The effects of these problems manifest as choppy audio. and variable delay (jitter). missing sounds. Please see page 147 for more details. lost packets.

• Low-latency queuing (LLQ): L L Q extends the C B W F Q system with the addition of a PQ. the admin classifies the traffic and assigns it to queues of configurable size and bandwidth allocation. This is the preferred queuing method for VoIP networks. The queuing strategies commonly used in Cisco Unified Communications include the following: • Weighted fair queuing (WFQ): W F Q dynamically assigns bandwidth to traffic flows as they arrive at the router interface. The PQ is typically reserved for voice traffic. all packets in the queue are immediately sent while packets of other traffic types are held in their respective queues. however. No configuration is necessary. VoIP needs a Priority queue (PQ) that guarantees it the bandwidth it needs. Realistically. The slowest link represents the available bandwidth for the entire path—often referred to as a bottleneck because of the congestion the slow link can cause. All rights reserved. Instead of the router dynamically interpreting traffic flows and building queues for them. . • Queuing: QoS employs advanced queuing strategies. and if any packets show up in the PQ. however. • Class-based weighted fair queuing (CBWFQ): C B W F Q extends the W F Q algorithm to include user-defined classes for traffic. which classify different traffic types and organize the classes into queues that are emptied in order of priority. This publication is protected by copyright. © 2008 Cisco Systems Inc. but instead allocates bandwidth fairly based on flow sizes (hence the name). If congestion is occurring. there are several ways to fix the problem: • Increase the bandwidth: If bandwidth is unlimited. at the expense of all other traffic. This strategy is not appropriate for VoIP because it does not provide a bandwidth guarantee for the voice traffic. congestion cannot occur. Please see page 147 for more details. That path may include a variety of L A N and WAN links. increasing bandwidth is costly and is usually unnecessary if QoS is applied instead. so C B W F Q is not appropriate for VoIP. There is still no priority queue.Bandwidth A VoIP call follows a single path from end to end. the strategy is enabled by default on router links of Tl speed and below.

Light travels just less than a foot in one-billionth of a second. which makes it appropriate for links that require the header to be readable to route the packet correctly (Frame Relay and ATM as examples). Please see page 147 for more details. so long-distance links can impose significant delay that cannot be eliminated. Sources of fixed delay include the following: • Propagation delay: The amount of time it takes for the signal to transit the link. © 2008 Cisco Systems Inc. Delay is classified in the following ways: • Fixed delay is predictable and constant. . This compression method does not affect the headers. All rights reserved. This is effectively the speed of light as it moves through copper or optical media. from 40 to as little as 2 bytes. • • Header compression: By specifying the use of compressed RTP (cRTP). T C P header compression is also available for non-VoIP traffic using T C P transport. to New York links routinely see 40 ms one-way propagation delay.A. link compression may be used. Delay Reducing end-to-end delay is a primary goal of QoS. Link compression: On point-to-point links where the header information is not needed to route the packet. Compression takes time and CPU resources. L. adding delay.• Compression: Several strategies are available to make the data that needs to be sent smaller so that it consumes less bandwidth: • Payload compression: By compacting the contents of a packet. the Layer 3 and 4 headers of a VoIP packet are dramatically reduced. This publication is protected by copyright. the total size is somewhat reduced. this must be factored in to the decision of what strategies are appropriate for a given link. Delay is calculated by adding the cumulative delay totals from source to destination and will be expressed as one-way or round-trip.

. Please see page 147 for more details. and many other factors that are not easily predictable or constant. • Overrun: Also the result of CPU congestion. this relates directly to the speed of the link and cannot be altered unless that speed is changed. © 2008 Cisco Systems Inc. no packets of any type will be lost. no more arriving packets can be placed in the queue. This is the most common source of packet loss. • Variable delay includes processing and queuing delays. arriving packets are dropped and lost. Employ appropriate compression techniques. for example). This is rare. Packet Loss Ideally.• Serialization delay: This is the time it takes to load bits onto the media. and it usually indicates an overloaded router CPU. overruns happen when the router cannot assign the packet to a free buffer space. the router performance. these will vary depending on the traffic load. Minimizing delay employs the same strategies as improving bandwidth: • • • Increase link speed. Packets are lost for a variety of reasons: • Tail drop: When an output queue is full. but this is not realistic. We do need to minimize packet loss for VoIP traffic because it has no mechanism to retransmit lost packets (unlike TCP. This publication is protected by copyright. All rights reserved. Use Priority queuing (such as LLQ) for delay-sensitive traffic. Any packets that arrive when the queue is full are dropped from the last position (tail) of the queue and cannot be recovered. • Input drop: If the input queue fills up.

shaping could limit the output to 256 kbps. it also requires another 150 bps per call for signaling traffic. All rights reserved. • Traffic policing: Drops packets in excess of a configured threshold. policing is an effective complement to QoS configurations. Again. These packets may be retransmitted if the traffic is using TCP. This is usually related to EMI or failing interface hardware. Minimizing loss can be achieved with QoS mechanisms like LLQ and compression or by increasing link speed. depending on the codec. and jitter for VoIP traffic. QoS Requirements for VoIP There are some accepted targets for delay. For example. © 2008 Cisco Systems Inc. delaying the transmission of the excess traffic.• Ignore: There is no buffer space available. loss. policing should not be applied to VoIP traffic. Some additional and complementary strategies known as Link Efficiency mechanisms will help to prevent congestion: • Traffic shaping: Delays packets and sends them at a configured maximum rate. Packet loss should be less than 1 percent. giant or runt frames. so it is not desirable to shape VoIP traffic. • Frame errors: Problems in transmission created CRC errors. compression. and Layer 2 in use. but shaping data traffic is an effective tool to complement voice QoS settings. This will add significant delay and might cause packets to be dropped. This publication is protected by copyright. Please see page 147 for more details. These are the targets that QoS and Link Efficiency mechanisms help us reach: • • • • Delay should be less than 150 ms one way. but because VoIP does not. if an F T P server is generating a 512 kbps stream. Jitter (the variation in the delay between packets) should be less than 30 ms one way. Each VoIP call requires between 17 kbps and 106 kbps of priority bandwidth. .

For example: a typical 384 kbps video stream should be allocated 460 kbps of priority bandwidth. it is appropriate to classify the enterprise data traffic into four or five classes and assign each a certain amount of bandwidth in its particular queue. The classifications you create will compose the QoS policy for the organization. Routine and Scavenger. Some apps will be characterized as critical to the business. through Mission Critical. and list all applications discovered in current use. and some as trivial or perhaps even undesirable. This will range from Priority for voice and video. Every business audit will generate a slightly different picture of what is vital to the business and what is to be disallowed.• The requirements for video are similar. Determine the level of service required for each app. which makes it simple to classify traffic that would otherwise be difficult or impossible. some as routine. This publication is protected by copyright. The Cisco QoS classification tools include Network-Based Application Recognition (NBAR). The policy will reflect the actual needs of both voice and data traffic on the network and will be a living document that will adapt to changes in the organization and the applications it uses for business. the ones used here serve only to identify the relative importance of the apps as they are classified. Determine whether congestion problems already exist. and a decision needs to be made about how to treat all applications discovered. It is not uncommon to discover that the business executive had no idea some apps were in use. A QoS policy is developed using the following process: 1. Perform a business audit to determine where the applications in use fit into the business model. and even Disallowed. Perform a network audit to determine the current state of traffic on the network. legitimate or otherwise. QoS Requirements for Data Traffic Although data requirements are not as strict as those of VoIP. Please see page 147 for more details. All rights reserved. . 3. 2. Urgent. The names are not important. © 2008 Cisco Systems Inc. bandwidth consumption is calculated as [video codec output] + 20 percent.

because generally we do not © 2008 Cisco Systems Inc. as appropriate. it may be re-marked and treated differently. All rights reserved. Ideally. AutoQoS generates traffic classes and service policies using predefined templates. Please see page 147 for more details. The autogenerated configuration adapts to changes (such as the relocation of an IP Phone) and is manually customizable to meet specific requirements after the automated config is completed. AutoQoS is a feature available on voiceenabled IOS platforms to greatly simplify and automate QoS configurations. the packet is treated according to the QoS marking and corresponding policy. This may include a minimum and maximum bandwidth allocation. This publication is protected by copyright. queuing strategies. both low. Use the business audit and the executive's decisions to create a classification scheme that lines up with the business needs. If it is not trusted. and link efficiency methods. This means that the trust boundary should actually be between the IP Phone and the attached PC. a priority for each class. Although it is essentially simple in architecture. the many commands needed are intimidating and time consuming. If it is trusted. AutoQoS QoS configuration is one of the more advanced skills in the IOS CLI environment. . Build the classification scheme to match the audit findings. Define the QoS settings for each traffic class. Router AutoQoS is limited to the following interfaces: • • • Serial PPP or H D L C links Frame Relay point-to-point links only ATM PVCs. making an in-depth understanding of the commands unnecessary. Auto QoS is available on all voice-enabled routers and switches with the correct IOS feature set. In any environment where there is a lack of skill or time. 5. we want to place the trust boundary as close to the source (the devices generating the traffic) as possible.and high-speed QoS Trust Boundary One of the important concepts in QoS is the trust boundary—the point at which the QoS marking of a packet or frame is believed by the switch or router. AutoQoS is a benefit.4.

when no phone is detected. All rights reserved. On a switch interface. The AutoQoS configurations are based on the configured bandwidth of the interface when AutoQoS is first run. the markings are not trusted. AutoQoS can automatically detect and configure the trust boundary by sensing a connected Cisco IP Phone and applying the necessary QoS commands. but we do trust the phone. lowering the bandwidth after AutoQoS is run will not change the AutoQoS configurations. the QoS marking of packets is trusted. so AutoQoS must be removed and reapplied if the bandwidth statement is changed. The switch must be able to recognize and configure the trust boundary. The fr-atm keyword is used on Frame Relay and ATM pointto-point links. we must move the boundary up to the gateway router. © 2008 Cisco Systems Inc. if it cannot. traffic is classified using NBAR. When a phone is detected. Using the [trust] keyword on a switch interface causes the inbound QoS marking of packets to be trusted regardless of whether a phone is detected. If trust is not configured. the keyword [ciscophone] enables the trusted boundary feature when the switch detects a Cisco phone through its CDP messages.trust the PC. If there is no phone. the trust boundary is between the switch and the PC. The keyword trust causes the DSCP markings of packets to be trusted for classification purposes. This publication is protected by copyright. Configuring AutoQoS The single command auto qos voip [trust] [fr-atm] enables AutoQoS at the interface. and the packets are D S C P marked as appropriate. Please see page 147 for more details. .

Unity Express. Please see page 147 for more details. you can specify it as single line (the default) or dual line. including the 2800 and 3800 series.com. Unified CM Express supports a wide range of TP Phone. and the ephone-dn is a destination number that can be assigned to multiple ephones. system. and conferencing features to work. . a dual line can terminate two calls at the same time. refer to the Unified Communications Manager Express 4. The optional GUI files may be installed for simplified configuration and administration but are not required. and third-party systems using H. In CM Express. The system extends the benefits of Unified Communications to small businesses. This is necessary for call waiting. Unified CM Express runs on the ISR platforms. along with IP Phone licenses and firmware.I n t r o d u c i n g Cisco U n i f i e d C o m m u n i c a t i o n s M a n a g e r Express Unified CM Express is a router-based call agent that scales up to 240 phones.323 or analog D T M F signaling. All rights reserved. the router automatically creates POTS dial peers to match. An ephone-dn always has a primary directory number. depending on platform capacity. an ephone is a logical configuration and settings for a physical phone. For a complete feature list. Unified CM Express supports all current-generation IP Phones. and trunk features. and on the 3700 series Multiservice Access Routers. This publication is protected by copyright. A single line can terminate one call. When you create an ephone-dn. as well as voice-mail integration with Unity. Defining Ephone and Ephone-DN An ephone is an Ethernet phone. and it may have a secondary one as well. The following © 2008 Cisco Systems Inc. The appropriate IOS IP Voice feature set. consultative transfer.2 Data Sheets on cisco. and flash and R A M appropriate for the install are required. and an ephone-dn is an Ethernet phone directory number. When you create an ephone-dn.

BBBB.e p h o n e ) # mac-address AAAA.e p h o n e ) # t y p e 7960 addon 1 7914 router(config-ephone)# button 1:20 Types of ephone-dns Six types of ephone-dns are configurable in CM Express: • Single line: This ephone-dn creates a single virtual port. It should be used when there is one phone © 2008 Cisco Systems Inc.configuration creates a dual-line ephone-dn with a primary and secondary number. this is controlled by the hardware capacity and by licensing. you should configure only what you will actually need. The max-dn <max-dn-value> command must be set to create ephone-dns.e p h o n e . . All rights reserved. The button command allows you to specify which button does what. Although you can specify a secondary number. The following configuration creates a basic ephone for a 7960 with a 7914 sidecar. Be aware that the router immediately reserves memory for the number of dns you specify. to which various functions can be applied. whether they are created or not. The number 20 in the configuration is the tag. Please see page 147 for more details. the phone can terminate only one call at a time. the top button is always numbered " 1 . CM Express will detect all phone models except the 7914 sidecar. The MAC address of the phone ties it to the ephone configuration. An ephone is the logical configuration of a physical phone.l i n e R o u t e r ( c o n f i g .d n ) # n u m b e r 5309 secondary 8675309 There is a maximum number of ephone-dns that a given platform will support. Each ephone is given a tag to uniquely identify it. which must be specified manually. which is simply a unique identifier: Router(config)#ephone-dn 20 d u a l . so it cannot support call waiting. the button 1:20 command assigns button 1 the dn (5309) assigned to ephone-dn 20 from the previous example: r o u t e r ( c o n f i g ) # ephone 20 r o u t e r ( c o n f i g . This publication is protected by copyright. " with the others following in numerical order.CCCC r o u t e r ( c o n f i g . Each different model of IP Phone has a different number of buttons. the default is zero.

call-park slots. • Shared ephone-dn: The same ephone-dn and number appears on two separate phones as a shared line. It should not be used for lines dedicated to intercom. meaning that either phone can use the line. transfer. or call park.or dual-line ephone-dn. M o H feeds. This is not a shared line because the phones will ring in succession. any one of the other phones sharing the line can pick it up. MWI. but only one can pick up. as explained in the "Hunting Configuration" section that follows. Router(config)#ephone-dn 1 Router(config-ephone-dn)#number 1001 • 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. but once in use the other cannot then make calls on that line. M o H feeds. . It can be either a single. Please see page 147 for more details. Router(config)#ephone-dn 2 d u a l . for example. and a call on hold can be answered only by the phone that placed it on hold. It should be used when a single button supports call features like call waiting. making it possible to dial two separate numbers to reach the phone. paging. This publication is protected by copyright. using three dual-line ephone-dns with the same number will terminate six calls on the phone. It can be used in combination with single-line ephone-dns on the same phone. • Multiple ephone-dns on one or more ephones: This configuration allows multiple calls to the same extension to be handled simultaneously on a single phone. and conferencing. it should be used when you want to have two numbers for the same button without using more than one ephone-dn. intercom.l i n e Router(config-ephone-dn)#number 1002 • Dual number: This ephone-dn has a primary and secondary number. It is useful for things like paging. If the call is placed on hold. © 2008 Cisco Systems Inc. and MWI. By using multiple ephone-dns on multiple phones. Controlling the hunting behavior (the order in which buttons or phones ring) is done with the preference and huntstop commands. all the phones can answer the same number. The line will ring on all phones that share the ephone-dn. All rights reserved.button for each PSTN line that comes into the system.

c o n fjg u r e s ephone-dns 30. The huntstop configuration sends calls to the first channel of ephone-dn 3. This can prevent calls from rolling over to the next ephone-dn. The following configuration creates an ephone with two ephone-dns that both terminate calls to 1003. 34. The preference command sets the order in which the call will be tried on a list of ephone-dns.35 and 35 on button 1 without call waiting. it is typical to send the call to voice mail.) The call coverage is similar to a shared-line setup.• Overlay ephone-dn: An overlay consists of two or more ephone-dns (up to 25) applied to the same button. then the second channel of ephone-dn 4. then the first channel of ephone-dn 4. with the result that all 10 phones could answer calls to the same number. All rights reserved. 3 1 . The overlay separator can be o. Router(config)#ephone-dn 3 d u a l . The command button lo30. . Please see page 147 for more details. which designates call waiting. The default is huntstop enabled. This is commonly used in environments where call coverage is needed to answer the same number. Hunting Configuration Hunting allows a call to search for an available line to ring.31>32. or c.33. This publication is protected by copyright. from this point. all these ephone-dns must be either single or dual line.l i n e Router(config-ephone-dn)#number Router(config-ephone-dn)#no 1003 0 Router(config-ephone-dn)#preference huntstop © 2008 Cisco Systems Inc. The button command with the overlay separator creates the overlay set. You can overlay up to 10 lines on a single button and then configure the same overlay set on 10 phones. then the second channel of ephone-dn 3. (You can't mix the types. which designates an overlay set without call waiting. except that a call to the number on one phone does not block the use of the same number on another phone. This causes the same phone to ring twice for the same call.34. so the no huntstop command must be used to allow the desired hunting behavior. 32. 33. such as a call center or help desk. If dual-line ephone-dns are configured. the huntstop command stops the hunting when it reaches that ephone-dn. the default behavior is for the call to hunt from the first line to the second.

We can force the call to hunt from channel 1 of the first ephone-dn directly to channel 1 of the second ephone-dn instead.CCCC 2:4 This is not necessarily the behavior we want.CCCC 3:6 Router(config)#ephone-dn 6 d u a l .BBBB. it is more common to use the second channel of an ephone-dn for transfer.BBBB. then to channel 2 of ephone-dn 6 (also on button 3): Router(config)#ephone-dn 5 d u a l .l i n e Router(config-ephone-dn#number 1004 0 channel Router(config-ephone-dn#preference Router(config-ephone-dn#huntstop Router(config-ephone-dn#number Router(config-ephone-dn#no Router(config)#ephone 4 Router(config-ephone#mac-address Router(config-ephone#button 2:5 AAAA. call waiting.l i n e 1004 1 channel Router(config-ephone-dn#preference huntstop In a call-coverage scenario.Router(config)#ephone-dn 4 d u a l . or conferencing. Please see page 147 for more details. Here we configure the call to hunt from channel 1 on the first phone to channel 1 on the second phone: Router(config)#ephone-dn 5 d u a l . using the hunststop channel command. . The following configuration will send the call from channel 1 of ephone-dn 5 (on button 2) to channel 1 of ephone-dn 6 (on button 3). All rights reserved.l i n e Router(config-ephone-dn)#number 1003 1 Router(config-ephone-dn)#preference Router(config-ephone-dn)#huntstop Router(config)#ephone 3 Router(config-ephone)#mac-address Router(config-ephone)#button 1:3 AAAA. we would want the call to hunt to an agent who is not already on the phone. This publication is protected by copyright.l i n e Router(config-ephone-dn#number 1004 © 2008 Cisco Systems Inc.

CCCC © 2008 Cisco Systems Inc. All rights reserved.l i n e 1004 1 channel Router(config-ephone-dn#preference Router(config-ephone-dn#huntstop Router(config)#ephone 4 Router(config-ephone#mac-address Router(config-ephone#button Router(config)#ephone 5 Router(config-ephone#mac-address Router(config-ephone#button 2:6 DDDD. .BBBB.EEEE. Please see page 147 for more details. This publication is protected by copyright.FFFF 2:5 AAAA.Router(config-ephone-dn#preference Router(config-ephone-dn#huntstop Router(config-ephone-dn#number 0 channel Router(config)#ephone-dn 6 d u a l .

loads or SCCPnn. 0 .x-y-x-w. it is only necessary to load the TERMnn. Filenames are case sensitive. This prompt is where your first steps of defining the max-ephones and max-ephone-dn settings (described earlier) would take place. The following is a sample command set: l o a d 7960-7940 P00303020214 l o a d 7920 c m t e r m _ 7 9 2 0 . . Telephony Service Configuration Manual setup of the CM Express system is done using the CLI. this is done using the load model firmware-file command. the tftp-server fiashifilename command is used. and the file extension should not be included in the command.loads firmware filename (without the .0 1 . Phone Firmware Loads The firmware files that were copied into Flash and made available to the phones via T F T P must be associated with the phones. Some phones require more than one file to be loaded—for example. although the other files must be available via TFTP. (Tip: Use the Cut-and-Paste function of your terminal client to prevent annoying typos!) For Java-based phones. the command telephony-service enables config-telephony mode.loads extension).C o n f i g u r i n g C M Express t o S u p p o r t Endpoints In this section we explore three methods of configuring endpoints on a CM Express system: configuring optional settings.com. For the router to serve the firmware to the phones. This publication is protected by copyright. Providing Firmware IP Phone firmware files ship with the CM Express software or can be downloaded from cisco. rebooting IP Phones. From the global config.x-y-x-w. and troubleshooting and verifying the configuration.7-0-3-0S © 2008 Cisco Systems Inc.0 8 l o a d 7941 TERM41. You must enter this command for every firmware file needed. Please see page 147 for more details. 4 . the 791 IG requires six separate files. All rights reserved.

The default SCCP TCP port is 2000 and does not normally need to be changed. the range is automatically registered if a gatekeeper is configured.) The command ip source-address ip-address [port port] defines the IP address of the router that will be used as the source for SCCP messages. for example. and port we just defined) and builds an X M L config file for each phone. The command is also needed to register the range of numbers the command specifies with a gatekeeper. The command no auto-reg-ephone prevents a phone from registering unless its MAC address is explicitly configured already.164 numbers to local extensions. but the option is available if the situation should require it. this allows a phone to be discovered and registered to an available ephone slot (provided the ip source-address command is configured). . the DIDs have the four-digit internal extension as the last four digits. DID Configurations It is common to have a range of DID numbers (fully qualified E. and one that you may repeat from time to time. usually. This publication is protected by copyright. the source IP address. The full syntax is dialplan-pattern tag pattern © 2008 Cisco Systems Inc. Autoregistration The autoregistration function is enabled by default.164 numbers) that allow outside callers to reach internal extensions directly. in fact. This function expands extension numbers to full E. (SIP signaling is also possible but is not covered in this document. once configured. Create XML Config Files The create cnf-files command takes the configurations (including the firmware load.Defining Source IP and Port The CM Express software uses SCCP to communicate with the phones. All rights reserved.164 numbers and converts incoming E. CM Express records the MACs of all phones that attempt to register but are blocked by autoregistration being disabled. CM Express supports this configuration with the dialplan-pattern command. Please see page 147 for more details. if you upgrade firmware or make other changes to the phone configuration. You can disable this with the no-reg keyword. use the show ephone attempted-registrations command to see the list and the clear telephony-service ephone attempted-registrations command to see and clear the list. This is a necessary step.

e x t e n s i o n . intercom or M W I functions. Nor can it be used for shared-line implementations. The auto assign start-dn to stop-dn [type phone-type] [cfw number timeout seconds] command syntax specifies the range of ephone-dns to use for a given phone model. Changes must be performed manually at the CLI. The 7914 sidecar is not supported by this command. The telephony-service auto-assign command will dynamically create ephones as physical phones are connected to the system. The following is a sample of how the command can be used: telephony-service a u t o a s s i g n 11 to 20 t y p e 7920 a u t o a s s i g n 21 to 30 t y p e 7940 a u t o a s s i g n 31 to 40 t y p e 7960 a u t o a s s i g n 41 to 50 ephone-dn 1 d u a l . any phone that registers will be assigned an ephone-dn from the specified range. it is desirable to automate the deployment of phones. The auto-assign cannot be used for ephone-dns that serve paging. The pattern uses the same wildcards as dial peers. You can enter multiple commands to specify ranges for your different phone types. You must have a range of ephone-dns configured. This publication is protected by copyright.p a t t e r n pattern [no-reg]. if no phone type is specified. but it is no longer necessary to create each ephone and associate it manually.l e n g t h 4 extension p a t t e r n 5 3 . Automated Deployment of Endpoints In some cases. A sample configuration to set up a dial plan pattern for extensions 5300-5399 and expand them to the DID range of 867-555-5300-867-555-5399 would look like this: telephony-service dialplan-pattern 1 86755553. assigning an available ephone-dn to the ephone.extension-length length e x t e n s i o n . single line or dual line). Please see page 147 for more details. MoH. phones with this add-on must have it manually added. . .l i n e number 5301 © 2008 Cisco Systems Inc. All rights reserved. followed by resetting the affected phones. the Call Forward Busy number to use (typically the voice-mail port).. and timeout values. The ephone-dns must all be the same type (that is.

All rights reserved. This publication is protected by copyright. 31 to 40 to 7960s. The user-locale languagecode command will change the language displayed on all 7940 and 7960 phones. and 41 to 50 to any other type of phone (including those already specified. 21 to 30 to 7940s. the 7920 is not affected and must be configured with its individual language capability local to the phone. if there are no more ephone-dns in their range). Please see page 147 for more details. Following are language codes supported for User Locale: • • DE: Germany DK: Denmark • ES: Spain • • FR: France IT: Italy • NL: Netherlands • • • • • • NO: Norway PT: Portugal RU: Russian Federation SE: Sweden US: United States (default) JA:Japan © 2008 Cisco Systems Inc. The network-locale language-code command will change the call progress tones and ring cadence (again with the exception of the 7920). time display. Location Customization CM Express supports phone display language. and ring cadence localization. .The preceding output assigns ephone-dns from 11 to 20 to 7920s.

.Following are language codes supported for Network Locale: • AT: Austria • • • • CA: Canada CH: Switzerland DE: Germany DK: Denmark • ES: Spain • • • • FR: France GB: United Kingdom IT: Italy JA: Japan • NL: Netherlands • • • • • NO: Norway PT: Portugal RU: Russian Federation SE: Sweden US: United States (default) To change the time display format. use date-format {mm-dd-yy I dd-mm-yy I yy-dd-mm I yy-mm-dd}. To change the date format. This publication is protected by copyright. Please see page 147 for more details. © 2008 Cisco Systems Inc. use the time-format {12 I 24} command. All rights reserved.

. This can be time consuming. Use reset when changing firmware. or at the config-telephony prompt to reset one or more phones. The reset command can be executed to reset a single phone at the config-ephone prompt. All rights reserved. The full syntax is reset {all [time-interval] I cancel I mac-address Isequence-all}. The command options work as follows: • all: Resets all phones. whether or not the reregistration of the previous has finished. the router waits 4 minutes as a timeout before resetting the next phone. © 2008 Cisco Systems Inc.Rebooting IP Phones There are two commands available to reboot IP Phones. • cancel: Stops the reset process. with the command parameters the same as the reset command. and speed-dial modifications. The TFTP Server IP should be the CM Express router. lines. • mac-address: Resets a specific phone. such as buttons. Troubleshooting Endpoints Check the following when troubleshooting: • Verify IP addressing: Use the Settings button on the phone to check the configuration of the IP phone. This command can also be executed either at the config-ephone prompt or at the configtelephony prompt. Please see page 147 for more details. each with a slightly different behavior. user/network locales. or URLs. The syntax is restart {all [time-interval] I mac-address}. The reset command causes a hard reboot of the phone and invokes D H C P and TFTP. The restart command causes a soft (warm) reboot and is useful for minor configuration changes. • sequence-all: The router waits for one phone to reset and reregister before resetting the next phone to prevent the phones from overloading the T F T P server. This publication is protected by copyright. • time-interval: Changes the interval between the router resetting the phones in sequence (default = 15sec).

. All rights reserved. • Verify the firmware installation of the phones: Use the debug ephone register command to verify which firmware is being installed. • Verify the phone setup: Use the show ephone command to view the status of the ephones and whether they are registered correctly. • Verify that the locale is correct: Use the show telephony-service tftp-bindings command to view the files that the TFTP server is providing. • Debug the T F T P server: Use debug tftp events to ensure that the Cisco Unified Communications Manager Express router is correctly providing the firmware and X M L files.• Verify the files in flash memory: Verify that the correct firmware files are in the flash memory of the Cisco Unified Communications Manager Express router using the show flash command. This publication is protected by copyright. © 2008 Cisco Systems Inc. • Review the configuration: Use the show running-config command to verify the ephone-dn configuration. Please see page 147 for more details.

This publication is protected by copyright. if you need to change this default multicast address (typically you do not).wav file to Flash (avoiding copyright issues by using royalty-free recordings). First copy a .wav in config-telephony mode. Please see page 147 for more details. Next. Music on Hold No one likes to be on hold. © 2008 Cisco Systems Inc.4. All rights reserved. but having something to listen to makes it a little better and can even relay useful information to the listener. Configuring Music on Hold (MOH) in CM Express is simple. issue the command moh wavefilename. By default. issue the command multicast m o h ip-address port port-number. .I m p l e m e n t i n g Basic V o i c e F e a t u r e s A business phone system is expected to provide the following features: • • • • • • • • • Music on Hold Call Forward Call Transfer Call Park Intercom Paging Call Pickup Call Blocking Directory Services The next sections describe the configuration of these basic business telephony features in CM Express.23. the router will multicast the stream to 239.10:2000.

not commonly used. Please see page 147 for more details. Using the CLI. for example. All rights reserved. the administrator can configure how this transfer happens using the transfer-system {blind I full-blind I full-consult llocal-consult} config-telephony command. • Full-blind: Calls are transferred immediately using the H. • call-forward maxlength length: Restricts the number of digits specified for the call-forwarding number.Call Forward The user can configure call-forwarding of all calls using the phone softkey. you can configure different call-forwarding options at the config-ephone-dn prompt: • • call-forward all directory-number: Forwards all calls to the specified directory number. this prevents call forwarding to an international long-distance number. © 2008 Cisco Systems Inc. • Full-consult: Calls are transferred with consultation (meaning the user must speak to the target of the transfer before the call is released). uses the H. • call-forward noan directory-number timeout seconds: Forwards calls to the specified directory number if the user does not answer the phone before the specified timeout.450.2 standard. • Local-consult: Uses a proprietary transfer method. The command options are as follows: • Blind: Calls are transferred immediately using a Cisco-proprietary method. .2 standard. call-forward busy directory-number: Forwards calls to the specified number if the user is on the phone.450. Call Transfer Users can transfer calls with the Transfer softkey. This publication is protected by copyright.

If the timeout keyword is not used. The range is from 0 to 65535. the Call Park reminder sends a 1-second ring and displays a message on the L C D panel of the Cisco IP Phone that parked the call and to any extension that is specified with the notify keyword. All rights reserved. • notify extension-number: (Optional) Sends a reminder ring to the specified extension in addition to the reminder ring that is sent to the phone that parked the call. reserved-for: (Optional) Indicates that this slot is a private park slot for the phone with the specified extension number as its primary line. • limit count: (Optional) Sets a limit for the number of reminder timeouts and reminder rings for a parked call. • only: (Optional) Sends a reminder ring only to the extension that is specified with the notify keyword and does not send a reminder ring to the phone that parked the call.Call Park Call park allows a user to hold a call but retrieve it from another location by dialing the call park extension. For example. The syntax is relatively complex and specifies several options: • park-slot [reserved-for extension-number] [timeout seconds limit count] [notify extension-number [only]] [recall] [transfer extension-number][alternate extension-number][retry seconds limit count]. By default. • recall: (Optional) Returns the call to the phone that parked it after the timeout limits expire. the reminder ring is sent only to the phone that parked the call. This publication is protected by copyright. A call-park extension is a "floating" ephone-dn that is not assigned to any ephone. The limit range is from 1 to 65535 reminders. Please see page 147 for more details. This option allows all reminder rings for parked calls to be sent to the phone of a receptionist or an attendant. a call parked at this slot is disconnected after the limit has been reached. . All lines on that phone can use this park slot. • • timeout seconds: (Optional) Sets the Call Park reminder timeout interval. calls are picked up in the order in which they were parked. Multiple calls can be parked at a single extension and are retrieved by dialling the extension. When the interval expires. in seconds. © 2008 Cisco Systems Inc. When a limit is set. a limit of 10 sends 10 reminder rings to the phone at intervals that are specified by the timeout keyword. for example. no reminder ring is sent to the extension that parked the call.

the parked call is automatically transferred to 5309.s l o t t i m e o u t 1 0 l i m i t 1 0 r e s e r v e d . • retry seconds: (Optional) Sets the delay before another attempt to recall or transfer a parked call. After a call has been parked for 100 seconds.f o r 5301 ephone-dn 13 number 7003 p a r k . it goes to 5310.s l o t t i m e o u t 1 0 l i m i t 1 0 t r a n s f e r 5309 a l t e r n a t e 5310 ephone-dn 12 number 7002 p a r k . The number of attempts is set by the limit keyword. A call parked in these slots will stay parked for 100 seconds and will send a notification every 10 seconds to the extension that parked it.• transfer: (Optional) Returns the call to the specified number after the timeout limits expire. if 5309 is busy or does not answer.f o r 5302 © 2008 Cisco Systems Inc.s l o t t i m e o u t 1 0 l i m i t 1 0 t r a n s f e r 5309 a l t e r n a t e 5310 ephone-dn 11 number 7001 p a r k . The range is from 0 to 65535. . If the 100-second limit elapses. • alternate: (Optional) Returns the call to a specified second target number if the recall or transfer target phone is in use on any of its extensions (ringing or in conversation). respectively. Please see page 147 for more details. All rights reserved. in seconds.s l o t t i m e o u t 1 0 l i m i t 1 0 r e s e r v e d . it will be disconnected. Ephone-dn 12 and 13 are reserved for 5301 and 5302. This publication is protected by copyright. ephone-dn 10 number 7000 p a r k . Ephone-dn 10 and 11 can be used by any extension. The following example creates four call-park slots.

Any user could dial the intercom if the extension is known.Intercom An intercom is a one-way audio speed-dial. but because they are no longer on the phone itself. . The following configuration shows a typical intercom configuration. B. All rights reserved. The destination phone answers the call in muted speakerphone mode so that privacy is maintained. users cannot dial them. it is still possible to configure them in an ephone-dn. These characters were at one time part of the touchtone dialpad. C. To make it impossible for anyone to dial the intercom (except those phones configured to do so). it allows the user to press a phone button and be directly connected to another user. This publication is protected by copyright. the extension number of the intercom can include the A. using the B digit as part of the intercom extension number: ephone-dn 10 number 5301 name "Tommy TuTone" ephone-dn 20 number 5309 name "Jenny" ephone-dn 51 number B5555 name "Tommy TuTone" i n t e r c o m B5556 l a b e l "Tommy TuTone" ephone-dn 52 number B5556 name "Jenny" i n t e r c o m B5555 l a b e l ephone 6 button button 1:10 2:51 1:20 2 : 5 2 ephone 7 "Jenny" © 2008 Cisco Systems Inc. Please see page 147 for more details. or D character. Commonly used by an executive to an admin assistant. however.

d n 25 ephone 2 © 2008 Cisco Systems Inc. 1 . or combined groups of phones. This publication is protected by copyright. Note the unicast keyword. which will override the multicast configuration if the phone is not reachable by multicast: Router(config-ephone)# paging-dn paging-dn-tag [unicast] The following example sets up a single paging group: ephone-dn 25 number 2525 name Paging S h i p p i n g p a g i n g i p 2 3 9 . The default transport is unicast. which limits paging to a maximum of 10 targets. All rights reserved. . multicast is also supported. 0 . 2 5 p o r t 2000 ephone-dn 18 number 1818 ephone-dn 15 number 1515 ephone 1 mac-address AAAA. When a user dials the paging extension. a group of phones. A paging group is created by configuring a dummy ephone-dn with the paging command and associating that ephone-dn with one or more ephones using the paging-dn command.BBBB.Paging Audio paging builds a one-way audio path from the speaker to a single phone.CCCC button 1:18 p a g i n g . The command syntax to create the ephone-dn is the following: Router(config-ephone-dn)# paging [ip multicast-address port udp-port] The following shows the syntax for associating an ephone to the paging ephone-dn. all configured phones answer the call in muted speakerphone mode. Please see page 147 for more details.

CCCC. the user must press the GPickup softkey and enter the group number of the ringing extension. the group numbers 81 and 817 will both be interpreted If the ringing extension is in the user's group.mac-address button 1:15 p a g i n g . paging-dn-tag. If the ringing extension is in another group. without belonging to a pickup group. If only one pickup group is defined. • Group pickup: A user can pick up a call for another group if the user knows the group extension. whether or not they are a member of a pickup group. or simply to combine other groups for paging. This publication is protected by copyright. but the leading characters must be unique to each group. users need only press the Pickup softkey.DDDD Combining Paging Groups The config-ephone-dn command paging-grouppaging-dn-tag. An ephone-dn is assigned to a pickup group with the command pickup-group number. for example. This is useful to create a paging group that reaches all phones for emergency use. is used to create a combined paging group from multiple. .. pressing the Pickup softkey will redirect the call to the user's phone.. The numbers are arbitrary. Call Pickup There are three variations of call pickup: • Directed call pickup: Any extension can pick up a call that is on hold on another directory number.. • Local group pickup: Users can pick up a ringing extension in their own group using the Pickup softkey plus the star key (*). Please see page 147 for more details.d n 25 BBBB. © 2008 Cisco Systems Inc. All rights reserved. previously defined paging dns.

h o u r s day wed 1 7 : 0 0 0 8 : 0 0 R o u t e r ( c o n f i g .h o u r s day sun R o u t e r ( c o n f i g .h o u r s date jan 1 j u l 4 00:00 00:00 © 2008 Cisco Systems Inc. and the holidays for New Year's day. .t e l e p h o n y ) # a f t e r . All rights reserved.t i m e When the after-hours schedule is in place.t e l e p h o n y ) # a f t e r .h o u r s day t h u R o u t e r ( c o n f i g . Using the 7-24 keyword blocks the configured pattern 24 hours a day. configurable with a PIN for authorized users. to 5:00 p.t e l e p h o n y ) # a f t e r . The following configuration sets up a call blocking plan for all calls outside of normal business hours of 8:00 a. but phones can be exempted from call blocking individually.t e l e p h o n y ) # a f t e r .h o u r s day day start-time stop-time a f t e r .t e l e p h o n y ) # a f t e r .h o u r s day mon 1 7 : 0 0 0 8 : 0 0 R o u t e r ( c o n f i g . An override function exists.h o u r s day t u e R o u t e r ( c o n f i g . and Christmas day: Router(config)#telephony. You can define up to 32 patterns of digits to block and apply a time schedule to restrict calls to whatever schedule suits your needs. This publication is protected by copyright.m.t e l e p h o n y ) # a f t e r . and disables the override PIN functionality. Monday through Sunday. use the block command to activate call blocking: after-hours block pattern t a g pattern [7-24] The patterns use the same syntax as dial plan patterns.t e l e p h o n y ) # a f t e r .h o u r s day sat R o u t e r ( c o n f i g .service R o u t e r ( c o n f i g .t e l e p h o n y ) # a f t e r .h o u r s date 17:00 08:00 17:00 08:00 17:00 08:00 17:00 08:00 17:00 08:00 00:00 00:00 R o u t e r ( c o n f i g . 7 days a week.h o u r s d a t e month d a t e s t a r t .h o u r s day f r i R o u t e r ( c o n f i g . Please see page 147 for more details.t i m e s t o p .Call Blocking Call blocking prevents unauthorized use of phones.t e l e p h o n y ) # a f t e r .m. Call blocking applies to all IP Phones (except analog FXS phones). The schedule can be configured by day or by date using the following config-telephony commands: a f t e r .. typically to specific number patterns or times of day. the Fourth of July.

t a g number name name] | clear} © 2008 Cisco Systems Inc. Directory entries are drawn from the ephone-dn configuration if it includes a name entry. All rights reserved. To create such an entry.t e l e p h o n y ) # a f t e r .. . The configtelephony command to specify how names in the directory shall be displayed is the following: directory {first-name-first | last-name-first} It is possible to configure a directory entry that is not an IP Phone. The directory entries can be listed as first-name-first or last-name-first..R o u t e r ( c o n f i g .h o u r s d a t e dec 25 0 0 : 0 0 0 0 : 0 0 Router(config-telephony)#after-hours block pattern 9011! Router(config-telephony)#after-hours Router(config-telephony)#after-hours Router(config-telephony)#after-hours Router(config-telephony)#after-hours Router(config-telephony)#after-hours block block block block block pattern pattern pattern pattern pattern 9011!# 91[2-9]. Directory Services Users can access the list of numbers and names by pressing the Directory key. CM Express supports 100 directory entries of up to 32 characters. This publication is protected by copyright.[2-9] 91900 91976 9[2-9].[2-9] Exempting a phone from after-hours blocking is easily configured with the after-hours exempt command at the configephone prompt. the name under the ephone-dn configuration should match to avoid confusion. whichever method is chosen. with the name being up to 24 characters. use the following configtelephony command: d i r e c t o r y e n t r y { [ e n t r y . Adding a PIN is equally simple at the same prompt with pin pin-number. Please see page 147 for more details.

M a i n t a i n i n g a CM Express S y s t e m An IP Phone system needs regular attention to watch for unusual events or unhealthy trends. of course. Whether the files are upgrades to the Cisco IOS. phone firmware or M O H files. This single file can be extracted on the TFTP/FTP server and the files downloaded to the router Flash memory. This publication is protected by copyright. the Communication Manager Express application or GUI. add features.) FTP is also supported if file sizes greater than 32 MB are to be moved. Managing Router Files The CM Express router will need routine updates applied to improve reliability. Please see page 147 for more details. or enhance security. . (The TFTP server must be active and accessible over the network.zip file containing all the files needed to run CM Express. © 2008 Cisco Systems Inc. All rights reserved. CM Express software is available as a bundled. single . Day-to-day operations and maintenance tasks include the following: • • • • Updating files on the router Configuring syslog logging Billing procedures Managing Call Detail Records (CDRs) The next sections discuss these topics and configurations. the command to load them into the router is the familiar copy tftp flash syntax. including the GUI. a suitable account and password must be configured for F T P transfers.

© 2008 Cisco Systems Inc. Please see page 147 for more details. .SYSLOG and SNMP MIB Support CM Express supports type 6 Syslog messages for IP Phone registration. using the Syslog viewer of your choice. along with CDR data. This publication is protected by copyright. After a Syslog server is configured using the logging ip_address command and is available on the network. these messages can be viewed along with other messages generated by the router. allow visibility into detailed information about both summary and specific call information. CM Express support for S N M P MIBs specific to IP telephony activities and events includes the following three MIBs: • • • Cisco-DIAL-CONTROL-MIB (CDR and call history) Cisco-VOICE-CONTROL-MIB (extends to telephony and VoIP dial peers and call legs) Cisco-VOICE-IF-MIB These MIBs. All rights reserved. The following are the Syslog messages provided for IP Phone registration events: • • • • • %IPPHONE-6-REG_ALARM %IPPHONE-6-REGISTER %IPPHONE-6-REGISTER_NEW %IPPHONE-6-UNREGISTER_ABNORMAL %IPPHONE-6-UNREGISTER_NORMAL S N M P allows network system administrators to monitor changes and events by way of messages sent to a monitoring application.

.323 start/stop time A A A messages to the syslog server or billing application. © 2008 Cisco Systems Inc. CDRs can optionally be sent to the Syslog server. The account code can then be accessed by a billing application to determine how long a user was on the phone with each customer. and billed accordingly. the account code is recorded in the C D R and added to the Cisco-VOICE-DIAL-CONTROL-MIB. All rights reserved. Please see page 147 for more details. If the user enters an account code using the Acct softkey during call setup or when the call is connected.Billing Support Billing support is provided by way of CDR records and H. This publication is protected by copyright. Call Detail Records Call Detail Records (CDR) are created by default and recorded in memory for later review and analysis using either the CLI or GUI.

All rights reserved. eliminating much of the administrative overhead. This publication is protected by copyright. Sessions Internal Card Storage Device Hours of Storage CUE—AIM CUE—NM CUE—NM-Enhanced 50 100 250 4 or 6 8 16 Yes No No Flash HDD HDD 14 300 300 Unity Express includes both a GUI and TUI interface. for initial mailbox setup as well as ongoing maintenance. . Please see page 147 for more details. Unity Express is actually an embedded Linux operating system. Cisco Unity Express supports up to 250 mailboxes (and 300 users). It can provide voice-mail and integrated messaging. Having a local voice-mail application is ideal for smaller organizations as a standalone solution or to provide local voice-mail access in a branch office of a larger organization without having to send the traffic across the IP WAN if bandwidth utilization is an issue. The TUI includes a tutorial to make it simple for users to set up their own mailboxes (or General Delivery boxes for group accessible mailboxes). dependent on hardware platform.) The following table summarizes the capacities of the three Unity Express hardware platforms. Mailboxes Max.I m p l e m e n t i n g Cisco U n i t y Express Unity Express is a richly featured voice-mail and auto-attendant application that is coresident in the router in either a Network Module format or Advanced Integration Module format. but not Unified Messaging. (This interface is not visible physically or in the router configuration. Unity Express Hardware Capacities Cisco Unity Express Module Max. and there is no provision for redundancy. There is no provision for a T D M interface to a legacy PBX voice-mail system (because the hardware is internal to the router). with an Ethernet interface to the router platform. Some of the features offered by Unity Express include the following: • • Alternate greetings—Allows a user to add a special greeting for an extended absence Message tagging (private or urgent) © 2008 Cisco Systems Inc.

This can be very useful. for instance. The Cisco Unity Express Editor is a tool that aids and speeds this process. but it goes beyond that by listening to the callers' responses to questions or options and offering more choices or playing specific greetings. or save messages Pause. and Time-ofDay and Day-of-Week call routing. If the Unity Express Editor GUI tool is not available and changes need to be made. options. Unity Express can run multiple AA scripts at the same time. Using the tool." you have heard an Auto Attendant.• • • • • • • Reply. Auto Attendant can eliminate the need for a receptionist—or at least free the person up to do other tasks. and responses. . Appuyez sur le 2 pour Francais. if the administrator © 2008 Cisco Systems Inc. All rights reserved. An Auto Attendant is a logical mapping of greetings. For many businesses. so that different greetings are played when the business is closed. providing for very flexible and detailed responses to customer calls. In addition. Creating one requires careful mapping of the decision and response tree. fast-forward. Using familiar Windows GUI-based actions. a TUI interface is also available. If you have ever heard "Press 1 for English. or rewind messages during playback Envelope information 0-to-Operator with definable destination extension for Operator Message Waiting Indicator (MWI) Mailbox Full notification VPIM compatibility for message interchange with other Unity Express systems (or any other VPIM-compliant system) Unity Express Auto Attendant An Auto Attendant is essentially an interactive answering machine. This publication is protected by copyright. forward. administrators can create multiple customized Auto Attendant flows. administrators can drag-and-drop steps into the AA tree. It answers incoming calls. Auto Attendant allows callers to search for the number of the person they are calling by first or last name. Please see page 147 for more details.

3.) A license file: Various license files can be loaded. setting up backup and restore operations.pkg provides 25 licenses. A command-line interface is also available for initializing the system and for times when the GUI is not available. although other languages are supported for the TUI and Auto Attendant. monitoring system resources (CPU. Certain tasks must be executed through the CLI. All rights reserved. memory).wakes up to a foot of new snow and has to call in to the system from home to record an emergency greeting that explains that the business is closed that day because of the snowfall. Unity Express GUI Much of the administration of UE can be managed from the administrative GUI. This includes normal operation such as setting passwords and PINs for users. upgrade and licensing.1.1. CUE provides the capability to bulk import users from Communications Manager Express at the command line or by using the GUI. Users also have access to a TUI that allows them to change their personal mailbox greetings and set or record their alternate greetings. if you must reload the software or perform upgrades.1: This is the installer file for version 3. The filenames include the number of licenses the file provides. creating users and groups. .1. (Other versions will have appropriate filenames.1. each allowing a specific number of mailboxes. however. and restarting the system. This publication is protected by copyright. and troubleshooting tasks such as viewing Syslog and trace files. setting up mailboxes. The TFTP server must hold the following files: • cue-installer. To use the GUI. Currently only English language support is offered for both GUI and CLI. for example. open the Unity Express U R L at http://module_ip_address from a supported browser. cue-vm-license_25mbx_cme_3. both a TFTP and an FTP server are required.1. Unity Express Software Files Unity Express comes preloaded with software from the factory.nm-aim. Please see page 147 for more details. These include software installation. • © 2008 Cisco Systems Inc.

0 Defines a software i n t e r f a c e Service-Enginel/0 interface ! T h i s is t h e CUE module p h y s i c a l i n t e r f a c e I ip unnumbered Loopback 0 © 2008 Cisco Systems Inc. some basic configurations must be applied to the host router: • Routing: Regardless of which routing protocol is in use.3.1.The FTP server must hold the following files: • cue-vm.nm-aim.1.3. 6 6 .prtl. cue-vm-en_US-lang-pack.nm-aim.255. the CUE router must be able to reach all networks that include hosts it must contact (voice-mail users. and so on).1. 1 ! i • 255.1.1. • cue-vm-full-k9.pkg: This is one of two system software files. the C U E module itself requires addressing to enable the "hidden" Ethernet link across the backplane. 1 6 8 . .255. In either case.1. IP addressing: In addition to any interfaces that require IP addresses for network connectivity. • cue-vm-installer-k9.nm-aim.1. All rights reserved. Please see page 147 for more details. Router Configuration Prerequisites Unity Express can be coresident in the CM Express router. or it can be installed in a different router. The recommendation is to set it up as follows: i n t e r f a c e Loopback 0 i p address 1 9 2 . call agents.3. This publication is protected by copyright.1.3.prtl: This is the second system software file.prtl: This is the application installation utility • A language file: For example.

2 I d e n t i f i e s t h e IP a d d r e s s of t h e CUE s e r v i c e .! i C o n f i g u r e s t h e module t o use t h e s o f t w a r e i n t e r f a c e I P ip address 192.66.0 service-module ! D e f i n e s t h e IP of t h e CUE o p e r a t i n g system I service-module ip ! ! route ip default-gateway 192.255. they are the two hosts in a dedicated.168.o f . All rights reserved. so a SIP dial peer with the following specific configurations must be created. • Create a SIP dial peer pointing at the C U E service-module.168.255. even though it is physically internal to the router.b a n d session dtmf-relay codec g711ulaw © 2008 Cisco Systems Inc. Please see page 147 for more details. C M E uses SIP to communicate with the C U E system. .2 255.255.p e e r voice ! ! 7000 v o i p 77.255 service-engine C o n f i g u r e s r o u t i n g f o r t h e CUE system t o reach t h e r o u t e r and t h e r e s t o f t h e network.66. even if there are no other SIP connections in the CM system: d i a l . C r e a t e s t h e v o i p d i a l peer Defines the d i g i t p a t t e r n of the mailboxes protocol sipv2 destination-pattern session ! ! ! Sets SIP as t h e p r o t o c o l used to communicate w i t h t h e CUE d i a l peer target ipv4:192.168.168.e n g i n e sip-notify Forces DTMF d i g i t s to be s e n t o u t .b a n d as SIP NOTIFY messages i n s t e a d of i n . "hidden" Ethernet link that exists only on the backplane between the C U E hardware and the software running on it. This publication is protected by copyright.66.66. The router sees the CUE module as a separate host. The Service-Engine and the Service-Module must be on the same subnet.1 1/0 192.255..2 255.

This means that if you use four digits for the local dial plan. . and that number of dots must equal the number of digits used in the local dial plan. . mwi on ephone-dn 76 number 4 4 7 6 . and the C U E admin © 2008 Cisco Systems Inc. . but in addition a pattern of dots must follow the digits. there must be five. All rights reserved. . There are some unique characteristics of these specialized DNs: the digit patterns should be unique in the system. and configure authentication: • • • Router(config)# ip http server: Enables the web server (it is disabled by default) Router(config)#ip http path flash: Sets the path to the http files as the root of the Flash directory Router(config)#ip http authentication {aaalenablellocalltacacs}: Sets the authentication type used when logging on to the web interface It is possible—and typically recommended—to define a Unity Express web interface administrator that is separate from the router administrator. and so on. define the path to the HTTP files. mwi o f f • Router H T T P access must be configured to support using the web-based GUI administration interface for both CUE and CME. if a five-digit plan is used. no vad • Configure ephone-dns for M W I on and off functionality. (The GUI for C M E is not covered in this document. . (All extensions in the local dial plan must use the same number of digits. . there must be four dots.[ 104] ! ! S e t s t h e codec to G . Often the router admin and the C U E system admin are not the same person. . This publication is protected by copyright.) The following commands will enable the HTTP server. w h i c h is t h e o n l y codec CUE s u p p o r t s I t i s recommended t o d i s a b l e VAD f o r t h e CUE system d i a l p e e r .) The resulting configuration will look something like this: ephone-dn 75 number 4 4 7 5 . 7 1 1 . of course. Please see page 147 for more details.

the CUE module starts automatically when the router is powered on.) Setting Up Unity Express After it is installed in the router. you are prompted with a message that all configuration info and data will be irrevocably deleted. This command will warn you that all calls will be terminated if you confirm with a "y. If you confirm "y. if the password is already M D 5 hashed. Command-line access to the C U E system is gained with the privileged exec command service-module service-engine mod/0 session. This publication is protected by copyright. . and AIM modules in particular are slower to boot than NM modules. To prevent the CUE admin from inadvertently causing unwanted changes to the router config. The module will generally take longer than the router to fully boot up. For remote access. the admin can create additional accounts for the customer administrator and users using the GUI.may not have the skills to administer the router. create a separate C U E web interface admin using the following command at the config-telephonyservice prompt: web admin system name username {password string | s e c r e t {0 | 5} string} The secret keyword encrypts the password in the router configuration." • restore factory default: Self-explanatory. All rights reserved." you have a factory-defaulted C U E system. connect to the router CLI using SSH and enter this command. © 2008 Cisco Systems Inc. (These accounts can also be created using the CLI. Please see page 147 for more details. use the secret 5 string command. After the initial C U E web admin account is created. If you want to enter a plain-text password that should be converted to an M D 5 hash. use the secret 0 string command. The following are some other useful C U E commands: • offline: Takes the CUE system offline. The exit command returns you to the router CLI.

t h i s p r o c e s s w i l l have c o n f i g u r e d installation configuration tool. IMPORTANT:: Do you w i s h to s t a r t Are you s u r e (y.n)?y configuration now (y. IMPORTANT: IMPORTANT: o r e n t e r t o use s e .n)?y t h e system w i l l b e h a l t e d IMPORTANT:: s o i t can b e s a f e l y removed f r o m t h e r o u t e r . IMPORTANT:: Once r u n . Please see page 147 for more details.0 .1 0 . (mydomain.1 0 ) o r e n t e r t o use l o c a l d o m a i n ) : E n t e r Domain Name Using l o c a l d o m a i n a s d e f a u l t © 2008 Cisco Systems Inc. All rights reserved. IMPORTANT:: t h e system f o r your l o c a t i o n . IMPORTANT:: IMPORTANT:: Welcome to C i s c o Systems S e r v i c e Engine IMPORTANT:: p o s t IMPORTANT:: IMPORTANT:: T h i s i s a one t i m e p r o c e s s w h i c h w i l l g u i d e IMPORTANT:: you t h r o u g h i n i t i a l s e t u p o f y o u r S e r v i c e E n g i n e . This publication is protected by copyright. The system will now ask you a series of questions to provide the basic information needed to allow it to interact with the network and let the administrator log in: E n t e r Hostname (my-hostname.com.9 0 .Post-Installation Configuration Tool A new (or factory-defaulted) CUE system will run the Post-Installation Configuration Tool the first time you log in. IMPORTANT:: IMPORTANT:: I f you d o not w i s h t o c o n t i n u e . .

90. Would you l i k e to use DNS f o r CUE ( y . Please s e l e c t a c o n t i n e n t o r o c e a n . 1 0 .1 or enter to bypass): E n t e r IP Address of t h e Secondary NTP S e r v e r (IP address.0. Please see page 147 for more details. . All rights reserved. E n t e r IP Address of t h e P r i m a r y NTP S e r v e r (IP address.90. 2 0 5 f o r s e r v e r s used by CUE. n ) ? n WARNING: WARNING: If DNS is not used CUE w i l l r e q u i r e t h e use of IP a d d r e s s e s . In o r d e r to c o n f i g u r e DNS you must know t h e IP address of at l e a s t one of your IMPORTANT:: DNS S e r v e r s .or server enter to bypass):10.0. © 2008 Cisco Systems Inc.1 Found 10.IMPORTANT:: IMPORTANT:: of IP a d d r e s s e s l i k e 1 . 1 0 0 . The next questions set the location and time zone for the C U E system: Please i d e n t i f y a l o c a t i o n s o t h a t t i m e zone r u l e s can b e s e t c o r r e c t l y . This publication is protected by copyright.

Arizona 9) M o u n t a i n Time . Please see page 147 for more details.s o u t h Idaho & e a s t Oregon . .most l o c a t i o n s 3) E a s t e r n Time .I n d i a n a .A l a s k a panhandle 14) A l a s k a Time 15) A l a s k a Time 16) A l e u t i a n .8) B o l i v i a 9) B r a z i l 10) Canada 11) Cayman 12) C h i l e Islands 25) Guyana 26) H a i t i 27) Honduras 28) Jamaica 29) M a r t i n i q u e 30) Mexico 31 ) M o n t s e r r a t 32) N e t h e r l a n d s 33) N i c a r a g u a Republic 34) Panama Antilles 42) Suriname 43) T r i n i d a d & Tobago 44) T u r k s & Caicos Is 45) U n i t e d 46) Uruguay 47) Venezuela 48) V i r g i n 49) V i r g i n Islands Islands (UK) (US) States 13) Colombia 14) Costa R i c a 15) Cuba 16) Dominica 17) Dominican #? 45 Please s e l e c t one o f t h e f o l l o w i n g t i m e zone r e g i o n s . All rights reserved.Navajo 10) M o u n t a i n S t a n d a r d Time 11) P a c i f i c Time 12) A l a s k a Time 13) A l a s k a Time .L o u i s v i l l e a r e a . This publication is protected by copyright.most l o c a t i o n s Islands © 2008 Cisco Systems Inc.Kentucky .Wisconsin border .west A l a s k a .M i c h i g a n 4) E a s t e r n S t a n d a r d Time 5) C e n t r a l Time 6) C e n t r a l Time . 1) E a s t e r n Time 2) E a s t e r n Time .M i c h i g a n 7) M o u n t a i n Time 8) M o u n t a i n Time .A l a s k a panhandle neck .

waiting 125 .. Please w a i t . .. Please see page 147 for more details. A d m i n i s t r a t o r Account C r e a t i o n © 2008 Cisco Systems Inc.17) Hawaii #? 11 The f o l l o w i n g i n f o r m a t i o n has been g i v e n : United Pacific States Time w i l l b e used. U n i v e r s a l Time is now: Mon Apr 28 1 7 : 0 1 : 2 0 UTC 2008. . you can l o g in to t h e C i s c o U n i t y Express GUI and run t h e i n i t i a l i z a t i o n w i z a r d . Therefore TZ='America/Los_Angeles' L o c a l t i m e is now: Mon Apr 28 1 1 : 0 1 : 2 0 MST 2008. This publication is protected by copyright. All rights reserved. With t h i s account. Is t h e above i n f o r m a t i o n OK? 1) Yes 2) No #? 1 C o n f i g u r i n g the system. . The next questions and answers create the administrator account and password: IMPORTANT:: IMPORTANT:: IMPORTANT:: IMPORTANT:: IMPORTANT:: IMPORTANT:: IMPORTANT:: E n t e r a d m i n i s t r a t o r user (user ID):UnityAdmin : E n t e r password f o r (password):Cisco ID: Create an a d m i n i s t r a t o r account.

FIGURE 12 Cue Initialization Wizard Login Screen © 2008 Cisco Systems Inc. All rights reserved. . Open a supported web browser and go to http://cue_ip_address/. Please see page 147 for more details. There are several links to choose from. you should be able to ping the IP address given to the C U E module from the PC you intend to use to administer it. This publication is protected by copyright. The first time you log in. we are going to examine the Initialization Wizard. a message displays stating that only Administrator logins are allowed (until other users have been configured on the system).C o n f i r m password f o r b y r e e n t e r i n g i t : (password):Cisco SYSTEM ONLINE CUE> At this point. CUE Initialization Wizard The Initialization Wizard allows you to quickly set up a brand-new (or factory-defaulted) C U E system.

FIGURE 13 Cue Initialization Wizard Entry Screen © 2008 Cisco Systems Inc.At the opening screen (see Figure 12). the message in red clearly indicates that the system has not been configured and that only Administrator logins are allowed. Please see page 147 for more details. Log in with the credentials you supplied earlier. This publication is protected by copyright. All rights reserved. .

FIGURE 14 Cue Initialization Wizard CM Login Screen <6» C i s c o CallManager Express > Powered bv-Cfscc -11 • .• 11 - CISCO Cisco Unity Express Initialization Wizard C a l l M a n j g t r Express Login Enter trie details of the CallManager Express that Cisco Unity Express will connect to The user name and password will be used to authenticate while retrieving information from the CallManagei Express Hostname': '101 10. All rights reserved. FIGURE 15 Cue Initialization Wizard Import Users Screen © 2008 Cisco Systems Inc. This publication is protected by copyright.2 User Name *: jCisco * indicates a mandatory field : . .| Next | : j Cancel | Help | The CM Express Login page lets you provide the address and credentials the CUE unit will use to contact the C M E router. Please see page 147 for more details. This is the IP address of the service engine.

user passwords and PINs. © 2008 Cisco Systems Inc.CUE will automatically import the users defined under the ephones in CME. You can then select whether users should have a mailbox. mailbox and message max size. FIGURE 16 Cue Initialization Wizard System Defaults Screen The next screen configures the system defaults for language. . Please see page 147 for more details. and whether to set CFNA and CFB. and message retention window. whether they are a voice-mail Administrator. This publication is protected by copyright. All rights reserved.

as well as defining M W I operation. Please see page 147 for more details. AA and the voice-mail operator. . This publication is protected by copyright. All rights reserved. © 2008 Cisco Systems Inc.FIGURE 17 Cue Initialization Wizard Call Handling Screen The Call Handling screen defines the DNs assigned for accessing voice mail.

The final screen lists the committed information. you are shown a review screen of the values you have entered so far. © 2008 Cisco Systems Inc. . This publication is protected by copyright. All rights reserved. Please see page 147 for more details. and you're given the option to commit the changes or go back to modify them.FIGURE 18 Cue Initialization Wizard Commit Screen Next.

All rights reserved. An AA can deal with multiple calls at the same time. Please see page 147 for more details. a series of recorded messages and interactive prompts allows you to create an answering system that gets callers either to an individual or to a voice mailbox so they can leave a message. This publication is protected by copyright. .FIGURE 19 Cue Initialization Wizard Committed Information Screen Auto Attendant The Auto Attendant (AA) is like the receptionist. One advantage of having an AA is that it is then possible to free up the receptionist to do other useful tasks. © 2008 Cisco Systems Inc.

FIGURE 21 AA Language Settings The first configuration is the language and script this AA will use. . © 2008 Cisco Systems Inc. Clicking the name will lead you to the configuration screens. This publication is protected by copyright. This shows a list of configured AAs. select Auto Attendant. All rights reserved. Please see page 147 for more details.FIGURE 20 Auto Attendant List Screen From the Voice Mail menu.

All rights reserved. FIGURE 23 AA Call Handling The Call Handling screen lets you specify the extension the system will dial to reach this AA and how many concurrent sessions the AA will support. . This publication is protected by copyright. Please see page 147 for more details.118] FIGURE 22 AA Scripts The next screen allows you to choose the individual recordings that the script calls. © 2008 Cisco Systems Inc. It is also possible to record custom AA script recordings.

The SBCS comes in two form factors: A desktop or wall-mount unit for installations of up to 16 users and a rack-mount unit for 32-48-user deployments. voice mail. . The SBCS incorporates C M E 4.2 and C U E 3. security.1. PoE switchports. AA. All rights reserved. and PSTN options are available. manageable. and secure wireless connectivity for both data and voice endpoints. Internet. WAN and PSTN connectivity options. including PoE switches. VPN. the smaller units support ISDN BRI PSTN. both PRI and CAS. and wireless options. This multiservice appliance incorporates routing. Hardware Components The core of the SBCS is the UC 500 Series for Small Business. video. including video and wireless capabilities. and wireless for up to 50 users. These all-in-one devices support data. and a simple-to-use graphical interface configuration tool makes it cost effective for small businesses to take advantage of Cisco's Unified Communications products. the Cisco Mobility Express Solution with the Cisco 521 Wireless Express Access Point and the Cisco 526 Wireless Express Mobility Controller provide scalable. © 2008 Cisco Systems Inc. This publication is protected by copyright. and FXS connections. with the features found on larger ISR hardware.1. can integrate with the SBCS to further leverage the productivity gains offered by unified communications.I n t r o d u c i n g t h e C i s c o S m a r t Business Communications System The Cisco SBCS is a unified communications appliance aimed squarely at the small-business market. both from Cisco and third-party vendors. For more complex wireless deployments. Specialized applications. The Catalyst 520 switch allows for expansion of the system to support more endpoints than the UC500 core unit supports. Please see page 147 for more details. The SBCS supports a wide range of Cisco IP phones. IPS. F X O . Many connectivity modules for WAN. They leverage UC500 Series devices. to provide the expansion capability to scale to the maximum endpoint capacity. and the larger units add support for Tl and El interfaces. firewall. voice.

including the following: • • PBX mode or keyswitch mode System features • • • • • Language Date format System message System speed dials Network features • • • Voice V L A N D H C P scope settings IP addressing • • SIP Trunk settings Dial Plan settings • • • Extension length Outgoing call handling Incoming call handling • Voice-mail features • • Voice-mail pilot numbers Auto Attendant © 2008 Cisco Systems Inc. . This publication is protected by copyright. All rights reserved.Telephony Features The SBCS supports most of the features desired in a business phone system. Please see page 147 for more details.

This publication is protected by copyright. All rights reserved. © 2008 Cisco Systems Inc.• • • • • • • • • • • Voice features MOH Paging Intercom Hunt Group Call Pickup Caller ID Blocking Call blocking Call Park Conferencing Users • • Name Association with a device • Phone • • • • M A C address Extension number(s) Permissions Call Forward Additional features are documented online. Please see page 147 for more details. .

showing the devices discovered in the system. and 802. All rights reserved. or external 521 Series wireless APs can be connected. showing ports and their status. and maintain the SBCS devices. LEAP.Security Features The SBCS supports the Cisco IOS firewall. lx authentication. The SBCS systems provide full support for wireless security. and language files. the use of a Cisco 526 Wireless Express Mobility Controller for every 6 APs is required. WEP. configure. This publication is protected by copyright. Easy VPN Server and Remote. PEAP. NAT. phone firmware. The standalone administrative capability of the Cisco Configuration Assistant will support up to three connected APs. . allowing control of the following: • • • • Switching Telephony Wireless Security • Network services • Internet connectivity The GUI tool provides a network map view. For support of up to 12 APs. as well as voice VLANs with QoS. as well as a front-panel view of the SBCS system. Cisco Configuration Assistant The C C A is a powerful and simple GUI tool for administering the UC500 Series platforms. including WPA and WPA2. The C C A even allows drag-and-drop upgrades to IOS software. Please see page 147 for more details. © 2008 Cisco Systems Inc. This tool is used to deploy. The larger SBCS models do not support internal APs. Wireless Features The smaller SBCS can be ordered with an integrated wireless AP.

Implementing Smart Business Communications System Voice Features The SBCS is remarkably simple to use. and sets up default © 2008 Cisco Systems Inc. .CCNA Voice Quick Reference by Michael Valentine Introducing the Cisco Smart Business Communications System FIGURE 24 Cisco Configuration Assistant Topology View UC520 SEP000D29C0198C SEP0012D9FF3979 SEP001794627A1A SEP0017E06A3FCC FIGURE 25 Cisco Configuration Assistant Front Panel View Cisco Unified 5 0 0 Series For those who miss the CLI. All rights reserved. it is still possible to do all administrative tasks from the command line if you so desire. Please see page 147 for more details. in fact. enables the device to place and receive calls on the PSTN interface. it ships with a default configuration that automatically assigns extensions to phones as they are plugged in. This publication is protected by copyright.

configurations for the firewall. This is as close to a plugand-play phone system as it gets.) Click Next. the default configuration gives the SBCS the IP of 192. It is recommended. that you use the Device Setup Wizard to perform the initial setup. however. because it integrates a number of setup procedures that are otherwise widely dispersed throughout the application.1. From the drop-down menu. . The following steps detail how to use the Device Setup Wizard: FIGURE 26 The CCA Device Setup Wizard— Step 1 1. This publication is protected by copyright. choose the device you want to configure. © 2008 Cisco Systems Inc. C C A will autodiscover any U C 5 0 0 Series devices that are connected and generate a topology map. Install the Cisco Configuration Assistant on your administrative PC. power them down or disconnect them if they are. Please see page 147 for more details. wireless (if applicable). and telephony features.) When you run the software. Select a Device: With the C C A open. Click Next. Prepare the Device: Verify that no other devices are connected. NAT. it will ask for the IP address of the system to connect to. (The installer is available as a free download from Cisco.com. choose Setup.10. All rights reserved. 2. (Only devices in the UC500 Series will be available. VLANs. Device Setup Wizard.168.

7. 8. All rights reserved. 4. 9. The C C A will verify connectivity to the device. This publication is protected by copyright. if it is already powered up. then click Next. Connect Device to Your PC/Laptop: You must connect to one of the PoE ports with a straight-through Ethernet cable. © 2008 Cisco Systems Inc. Wait until your PC has obtained an IP address. This may take a minute or two. Verifying Connectivity: The CCA will contact the device and confirm connectivity to it. These settings change the ring cadence on the phones as well as the languages displayed and/or heard on the system. you select the WAN interface and can then choose to disable D H C P and set a static IP address.3. Power Up Device: You are prompted to power up the device. you select the Region. Phone Language. Summary: A brief summary of the configuration you have entered is displayed along with a brief caution that the update may take up to 10 minutes. 1 0 . and Voicemail language as appropriate to the device's location. Click Next. 6. you can skip this step and configure NTP later. Hostname and User Authentication: Enter the administrator username and password. 5. . Enter IP Address and Other Device Setup Parameters: In this screen. Enter Date and Time Information: You have the choice of synchronizing the time to the PC's clock or setting it manually. Please see page 147 for more details. If you want to use N T P for the device's time synchronization. Enter Other Device Setup Parameters: In this section. click Next.

FIGURE 28 The Setup Menu n $Q Selm Device Setup Wizard. you have access to the menus in the left pane. or create a new community of devices. if you have several customers. and wireless access controllers). Communities make centralized management of a related set of devices simpler. 520 Series switches. you could create a community for each customer. modify options for connection port numbers. A community is a group of SBCS devices (including 500 Series routers. each of whom has an SBCS system. under which is located the Device Setup Wizard detailed earlier.. making your administrative organization simpler. © 2008 Cisco Systems Inc. All rights reserved. The devices might not be in the same physical location or logical subnet. the Connect window appears.FIGURE 27 The CCA Connect Window When you launch the CCA application. Here you can enter a specific community. Please see page 147 for more details. or hostname of a device to connect to. . The first menu is the Setup menu. C C A Menus After connection to the device or community. wireless APs. for example. This publication is protected by copyright. IP address.

All rights reserved. Please see page 147 for more details.FIGURE 29 The Configure Menu Next is the Configure menu. Hostname: Allows you to change the hostname of a device. which has options to configure Ports. . This publication is protected by copyright. the submenus include the following: • • • IP Address: Allows you to view and change the IP addresses on a device and set DNS server IPs. • Device Access: Here you can set the allowed terminal protocols (Telnet. and Internet Connection. System Time: Allows you to view. set. or both) that can be used to access a device. Telephony. © 2008 Cisco Systems Inc. Security. Device Properties. Under Device Properties. • H T T P Port: Allows you to change the HTTP default port the device uses. • Users and Passwords: Here you can change the administrator password on a device or on all devices simultaneously. SSH. and sync the time as well as configure N T P settings on one or more devices. Routing. You can also save the system configuration from this menu.

. The Health link generates graphical charts showing the statistics for key performance counters. community strings. FIGURE 3 0 The Monitor Menu The Monitor menu allows you to generate reports and change the view from Front Panel to Topology. including location and contact. and restart or reset devices. Please see page 147 for more details. All rights reserved. The license management option visible here may not be supported by the UC500 IOS in use. and so on. manage your configuration archive. © 2008 Cisco Systems Inc. This publication is protected by copyright.S N M P : Here you configure S N M P settings. manage the files stored on device Flash memory. FIGURE 31 The Maintenance Menu The Maintenance menu gives you the ability to perform software upgrades (with drag-and-drop functionality). traps to send. The Event Notification and System Event Messages links allow you to view and acknowledge messages and resolve problems automatically using Cisco Configuration Assistant (if possible). depending on the model.

or a M A C address. Topology generates a network map of the SBCS devices discovered or named in the community. port IDs. Please see page 147 for more details.Topology View Under the Monitor menu. This view allows you to annotate devices with IP addresses. . selecting Views. Right-clicking or double-clicking a device allows you to view its properties or change the settings of devices. the options vary depending on the device selected. a friendly name. All rights reserved. Front Panel View © 2008 Cisco Systems Inc. This publication is protected by copyright.

this is seldom necessary because it is autodiscovered by CCA. The following explains what is found and configurable in each tab: FIGURE 34 The Voice Device Tab • Device: The Device tab allows you to modify the hardware configuration. and PoE settings of an Ethernet port. interactive representation of the device. All rights reserved. If there is an error or missing information on any page. however. For example. . (This decision is part of the planning process and will be largely made by the customer. Configure Menu: Telephony The Telephony menu includes the Voice configuration screen. This publication is protected by copyright. Please see page 147 for more details.Next under Views is the Front Panel View. the tab will be highlighted in red. you can choose to configure all ports in a module or a single one.) This screen also lists the number of licenses (IP Phones) the unit supports. © 2008 Cisco Systems Inc. this shows a graphical. which in turn includes several tabs. speed. Interfaces are configurable by right-clicking them. you can enable or disable the port and configure the duplex. Here we can change the call agent from a PBX to a Key system depending on the needs of the customer. with advice from the designer.

FIGURE 35 The Voice System Tab System: The System tab lets you configure region. All rights reserved. clock format. Please see page 147 for more details. and phone language settings. . This publication is protected by copyright. and system speed dials. © 2008 Cisco Systems Inc. voice-mail.

All rights reserved. Please see page 147 for more details. This publication is protected by copyright.FIGURE Tab 36 The Voice Network • N e t w o r k : This is where you configure the Voice V L A N and D H C P scope. . © 2008 Cisco Systems Inc.

as well as their PSTN access numbers. Please see page 147 for more details. This publication is protected by copyright. © 2008 Cisco Systems Inc. . All rights reserved.FIGURE 37 The Voice AA & Voicemail Tab • AA & Voicemail: Here you set the extension numbers for the Auto Attendant and Voicemail.

. as well as generic SIP trunks for other providers.FIGURE 38 The Voice SIP Trunk Tab • SIP Trunk: SIP trunks are used to connect to other telephony devices or service providers. This publication is protected by copyright. On this page you identify the SIP Proxy and Registrar servers and the M W I server. All rights reserved. The SBCS provides built-in support for AT&T and CBeyond Communications SIP trunking services. and define domain information. Please see page 147 for more details. define the digest authentication username and password. FIGURE 39 The Voice Features Tab © 2008 Cisco Systems Inc.

Choosing North American preconfigures the area codes as threedigit. you can adjust the number of digits per extension (the default is 3) and set the numbering plan locale to North American or Other.Voice F e a t u r e s : In this screen. This publication is protected by copyright. and the international code as O i l — t h e s e are all standardized as part of the North American Numbering Plan. This page also allows you to configure the behavior for incoming calls. the long-distance access code as 1. . FIGURE 40 The Dial Plan Tab • Dial P l a n : In the Dial Plan screen. Please see page 147 for more details. Hunt Groups. Additionally. Group Pickup. enable and configure Paging. you identify the M O H audio file. All rights reserved. Intercom. either send them to an operator or have calls on a particular FXO port sent to a specific extension. you can configure the CallerlD Block code and the Outgoing Call Block List. and Conferencing. Choosing Other allows you to customize the dial plan as needed for other numbering plans worldwide. Call Park. © 2008 Cisco Systems Inc.

All rights reserved. This publication is protected by copyright. Please see page 147 for more details. © 2008 Cisco Systems Inc.FIGURE 41 The Users Tab • U s e r s : Here you associate users with the phones discovered by CCA and add new phones as needed. . You also have access to the phone configuration screen by clicking the More button.

define paging group. a firewall. All rights reserved. select Ports. Implementing Additional Smart Business Communications System Features The SBCS includes support for many features beyond the telephone system. a D H C P server. Port Settings From the Configure menu.The More Screen In this screen. configure user permission. an Ethernet switch. it is also a router. Port Settings. and set timers and operations for busy and no-answer rules. you can change what the phone buttons do. This section will review the configuration of these elements. . This publication is protected by copyright. and optionally a wireless AP. Please see page 147 for more details. © 2008 Cisco Systems Inc. configure intercom.

This publication is protected by copyright. © 2008 Cisco Systems Inc.FIGURE 43 Port Settings Configuration The Configuration Settings tab (shown in Figure 43) allows you to enable and disable ports. and enable or disable PoE negotiation. . All rights reserved. set duplex and speed. Please see page 147 for more details.

here you see that ports have actually negotiated Full Duplex/100 Mbps and PoE. Cisco. it hides internal addresses from the outside network (typically the Internet). and Cisco being the prestandard proprietary implementation. and IEEE under the Device column. NAT Network Address Translation serves three purposes: First. (In contrast to the setting of Auto in the Configuration tab. © 2008 Cisco Systems Inc. Second. expressed as Consumed and Remaining values. The display shows Unknown. it can allow many internal addresses to access the Internet using a single. This publication is protected by copyright. Unknown typically means the attached device does not need PoE). All rights reserved.The Runtime Status tab shows what the port is actually doing. VPN Server.) At the top of the table you can see the allocated PoE. Security Under the Security menu. Please see page 147 for more details. Security Audit. and Firewall and D M Z . you will find submenus for NAT. registered Internet IP. . these relate to the different PoE delivery types (IEEE being the current standard.

from here. Security Audit The Security Audit link allows you to inspect and report on the security configuration of a particular device. Next. it can provide selective access to internal IPO addresses from the outside in a controlled manner. you can © 2008 Cisco Systems Inc. This publication is protected by copyright. . Third. Please see page 147 for more details. for example. You are presented with a list of security checks and an indication of whether the device has passed the check. this is commonly used if security is less of a concern. The option of enabling Split Tunneling allows clients to use their own Internet connection for any network other than the ones listed. VPN Server The V P N Server page lists and allows you to create the user accounts that can access the system via VPN (to a maximum of 10 concurrent sessions). this is useful for reaching mail and FTP servers from the Internet. You must define a preshared key. FIGURE 45 The NAT Page The NAT page allows you to configure these specific server targets. as well as firewall service configuration. which is used in the authentication and encryption process. define the IP address range that will be assigned to remote clients connecting to the system.These first two capabilities are enabled by default on the SBCS. All rights reserved.

define which interfaces are trusted and untrusted. The interface allows you to create a scope of addresses for each VLAN. such as PCs. subnet mask. and default gateway values to hosts on the LAN. and also to define which interface is the D M Z (Demilitarized Zone—a term that describes a screened network where certain servers and resources are placed so that controlled access to them can be provided without risking the private network). © 2008 Cisco Systems Inc. . Please see page 147 for more details. The SBCS D H C P server is suited to the task of a small network deployment and should not be used for larger environments. All rights reserved. If this is the case. Firewall and DMZ The Firewall and D M Z page allows you to configure the basic security level (High. DHCP Configuring a D H C P server allows the SNCS to allocate IP address. (A typical system will have one VLAN for the phones and at least one more for the data devices.) You can also configure static D H C P bindings (so that you can predict what IP a given MAC address will be assigned) and which addresses or range of addresses will be excluded from the D H C P scope. or Low) of the firewall to apply a preconfigured set of typical restrictions. until the best course of action to both resolve the security issue and allow the intended operation can be determined. you do have the ability to configure static routes to ensure the device can reach remote subnets not directly connected. be aware that increasing the security settings of a device may block connectivity to some applications.select one or more checks and click OK to have the C C A fix the security problem automatically. Routing Although the SBCS does not typically run dynamic routing protocols (being designed for smaller installations where such power is not required or will be handled by other devices). Although it is convenient and simple. the change can also be undone in this interface. Medium. This publication is protected by copyright.

You can also view the port configuration for the entire device by clicking its image and then clicking Details. Wireless If the SBCS is equipped with or connected to a wireless device. © 2008 Cisco Systems Inc. All rights reserved. The interface also allows you to view and set the Access (data) and Voice VLANs per port. .1Q trunking protocol.Smartports FIGURE 46 Smartports The Smartports feature allows for rapid configuration of common interface settings appropriate to different device types. by selecting Configure. for example. W L A N s you can view and change settings for the SSIDs for data and voice (for use with wireless IP Phones such as the 7920 and 7921). Please see page 147 for more details. Selecting an SSID allows you to view and configure the wireless settings for the SSID. V L A N : Change the V L A N to which the SSID belongs. selecting Switch or Router from the pull-down list associated with a port will activate the 802. including the following: • • B r o a d c a s t in B e a c o n : Select whether to make the SSID visible to wireless devices. selecting IP Phone + Desktop will configure multiple-VLAN functionality and QoS settings. This publication is protected by copyright.

with links to Front Panel and Topology (discussed previously). All rights reserved. © 2008 Cisco Systems Inc. Health. and System Messages. mask. specify the use of PPPoE if your Internet provider requires it. among others. which relies on the negotiation capabilities of PPPoE to determine an IP address. or WPA2. using WEP.• Security Settings: Change from the default of no security to a setting that may include authentication. encryption. D H C P can be used. Event Notification. and choose the addressing method. you can choose IP Negotiated. Views. If you have selected PPPoE. or both. This publication is protected by copyright. or if your service provider has allocated you a static IP. Internet Connection This screen allows you to view and change the settings for the WAN interface. Please see page 147 for more details. with links for Inventory and VPN Status. and default gateway. Save Configuration This simple screen allows you to save the configuration of one or all devices to N V R A M . LEAP. You can enable or disable the interface. WPA. . you can specify the IP. Maintaining a Smart Business Communications System Several tools are included in C C A to monitor and maintain the SBCS. The Monitor menu includes Reports. making it the startup configuration at the next reboot of the device.

This publication is protected by copyright. Packet Error Rate. Please see page 147 for more details. Temperature. More information can be read in the Health Details window.Health FIGURE 47 The Health Display Under the Monitor menu. PoE Utilization. This generates a graphical representation (shown in Figure 47) of the critical general health statistics of the SBCS: Bandwidth Utilization. . CPU Utilization. and Memory Utilization. select Health. © 2008 Cisco Systems Inc. accessed by clicking the Details button. These stats are updated every minute. All rights reserved.

Errors are marked as Level 2 or 3. Informational events are marked as Level 5. Please see page 147 for more details.Event Notification CCA allows you to view Event logs for all devices in your network. These messages are the same that can be seen at the terminal monitor of the IOS CLI. All rights reserved. 6. © 2008 Cisco Systems Inc. and turn off the Alert LED on SBCS switches. or 7. Warnings are marked as Level 4. . The Filter button allows you to view only messages of the selected level(s). if desired. System Messages This screen allows you to view system messages from all devices or any single device and apply filters for severity level. with easy-to-read severity icons to quickly indicate whether the event is serious or informational: • • • • Critical errors are marked as Level 0 and 1. tell CCA to take action where possible. This publication is protected by copyright. The icons for these messages are shown next: FIGURE 48 Event Monitor The Event Notification window allows you to acknowledge event notifications.

you can change the directory and choose akp to save the configuration to the device before backing it up.Backup and Restore From the Maintenance menu. the next link is Restart/Reset. note any descriptive comments. © 2008 Cisco Systems Inc. by selecting the Preferences button.configuration D c m ns a e\ assistants b c u s directory by default. To back up. and click Restore. Restart/Reset Under the Maintenance menu. . select a backup file. The Restore tab allows you to select your view of backed-up configurations: • • • Show backed-up configurations of the selected device Show backed-up configurations of the selected device type Show all backed-up configurations Choose a device to restore. select All Devices or just a single device. choose Configuration Archive. The window displays the Backup tab and the Restore tab. All rights reserved. Add a descriptive note for future reference. and click Back Up. This publication is protected by copyright. This allows you to reboot the chosen device and gives you the option of resetting it to factory defaults if need be. The files are written to the C: \ o u e t and Settings\<user n m > . Please see page 147 for more details. make note of the backup path.

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