Cisco ATA 190 Administration Manual

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Cisco ATA 190 Analog Telephone Adapter Administration Guide for SIP

Version 1.0

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Cisco ATA 190 Analog Telephone Adapter Administration Guide for SIP

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CONTENTS

Preface vii

Cisco ATA 190 Analog Telephone Adapter Overview 1-1

Session Initiation Protocol Overview 1-2

SIP Capabilities 1-2 Components of SIP 1-2

SIP Clients 1-3 SIP Servers 1-3

Hardware Overview 1-4

Software Features 1-4

Secure Real-Time Transport Protocol 1-5 Name Signaling Event based passthrough 1-5 Transport Layer Security Protocol 1-5 T.38 Fax Relay 1-5 Voice Codecs Supported 1-5 Other Supported Protocols 1-6 ATA 190 SIP Services 1-6 Modem Standards 1-7 Fax Services 1-7 Methods Supported 1-7 Supplementary Services 1-8

Contents

Installation and Configuration Overview 1-8

Preparing to Install the ATA 190 on Your Network 2-1

Understanding Interactions with Other Cisco Unified IP Communications Products 2-1

Understanding How the ATA 190 Interacts with Cisco Unified Communications Manager 2-2

Providing Power to the ATA 190 2-2

Power Guidelines 2-2 Power Outage 2-2

Understanding Phone Configuration Files 2-3

Understanding the ATA 190 Startup Process 2-4

Adding the ATA 190 to the Cisco Unified Communications Manager Database 2-5

Adding the ATA 190 with Auto-Registration 2-6 Adding the ATA 190 with Cisco Unified Communications Manager Administration 2-6

Determining the MAC Address of an ATA 190 2-7

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Contents

Installing the ATA 190 3-1

Cisco ATA Rear Panel Connections 3-1

Network Requirements 3-1

Safety Recommendations 3-1

What the ATA 190 Package Includes 3-2

Installing the ATA 190 3-2

Attaching a Phone to the ATA 190 3-3

Verifying the ATA 190 Startup Process 3-3

Configuring Startup Network Settings 3-3

Configuring Security on the ATA 190 3-3

Configuring the ATA 190 4-1

Telephony Features Available for the ATA 190 4-1

Configuring Product Specific Configuration Parameters 4-4

Adding Users to Cisco Unified Communications Manager 4-6

Configuring Fax Services 5-1

Using Fax Mode 5-1

Fax Modem Standards 5-1 Fax Modem Speeds 5-2

Using SIP Supplementary Services 6-1

Common Supplementary Services 6-1

Attended Transfer 6-2 Call Pickup 6-2 Caller ID 6-2 Call-Waiting Caller ID 6-2 Call Hold 6-2 Group Call Pickup 6-3 Meet–Me Conference 6-3 Privacy 6-3 Shared Line 6-3 Speed Dial 6-4 Redial 6-4 Unattended Transfer 6-4

Semi-unattended Transfer 6-4 Fully Unattended Transfer (Blind Transfer) 6-4

Voice Mail Indication 6-5

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Voice-Messaging System 6-5 Making a Conference Call in the United States 6-5 Making a Conference Call in Sweden 6-5 Call Waiting in the United States 6-6 Call Waiting in Sweden 6-6 About Call Forwarding 6-6 Call Forwarding in the United States 6-6 Call Forwarding in Sweden 6-6

ATA 190 Specifications A-1

Physical Specifications A-1

Electrical Specifications A-2

Environmental Specifications A-2

Physical Interfaces A-3

Ringing Characteristics A-3

Contents

Software Specifications A-3

SIP Compliance Reference Information A-4

Voice Menu Codes B-1

Accessing the IVR and Configuring Your Phone Setting B-1

Recommended ATA 190 Tone Parameter Values by Country C-1

Troubleshooting and Maintenance D-1

Resolving Startup Problems D-1

Symptom: The ATA 190 Does Not Go Through its Normal Startup Process D-1 Symptom: The ATA 190 Does Not Register with Cisco Unified Communications Manager D-2

Checking Network Connectivity D-2 Verifying TFTP Server Settings D-2 Verifying DNS Settings D-3 Verifying Cisco Unified Communications Manager Settings D-3 Cisco Unified Communications Manager and TFTP Services Are Not Running D-3 Creating a New Configuration File D-3 Registering the Phone with Cisco Unified Communications Manager D-4

Symptom: ATA 190 Unable to Obtain IP Address D-4

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ATA 190 Resets Unexpectedly D-5

Verifying Physical Connection D-5

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Contents

LOSSARY

NDEX

Identifying Intermittent Network Outages D-5 Verifying DHCP Settings D-5 Checking Static IP Address Settings D-6 Verifying Voice VLAN Configuration D-6 Eliminating DNS or Other Connectivity Errors D-6

Troubleshooting ATA 190 Security D-7

General Troubleshooting Tips D-7

Where to Go for More Troubleshooting Information D-9

Cleaning the ATA 190 D-9

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Overview

Audience

Preface

The Cisco Analog Telephone Adapter 190 Administration Guide for SIP provides the information you need to install, configure, and manage the Cisco ATA 190 Analog Telephone Adapter (ATA 190) on a Session Initiation Protocol (SIP) network.

This guide is intended for service providers and network administrators who administer Voice over IP (VoIP) services using the ATA 190. Most of the tasks described in this guide are not intended for end users of the ATA 190. Many of these tasks impact the ability of the ATA 190 to function on the network, and require an understanding of IP networking and telephony concepts.

Organization

This manual is organized as follows:

Chapter 1, “Cisco ATA 190 Analog Telephone Adapter Overview”

Chapter 2, “Preparing to Install the ATA 190 on Your Network”

Chapter 3, “Installing the ATA 190” Provides information on how to connect the

Chapter 4, “Configuring the ATA 190 for SIP Provides information on how to configure the

Provides descriptions of hardware and

are features of the ATA 190 along with a

softw brief overview of the Session Initiation Protocol (SIP).

Provides information on the interactions

tween the ATA 190, Cisco Unified

be Commu It also describes options for powering the ATA

190.

A firmware files.

A Protocol (SIP).

nications Manager and other devices.

TA 190 hardware and load the QED and

TA 190 to operate with Session Initiation

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Preface

Chapter 5, “Configuring and Debugging Fax Servi

ces”

Chapter D, “Troubleshooting and Maintenance”

Chapter 6, “Using SIP Supplementary Services”

Chapter B, “Voice Menu Codes” Provides a quick-reference list of the voice

Appendix A, “ATA 190 Specifications” Provides physical specif

Appendix B, “SIP Call Flows” Provides ATA 190 call flows for SIP scenarios.

Appendix C, “Recommended ATA 190 Tone

arameter Values by Country”

Glossary Provides definitions of commonly used terms.

Index Provides reference information.

Related Documentation

For more information about the ATA 190 or Cisco Unified Communications Manager, refer to the following publications:

Cisco ATA 190 Analog Telephone Adapter

• RFC 3261 (SIP: Session Initiation Protocol)

• RFC 2543 (SIP: Session Initiation Protocol)

Provides instructions for configuring both ports of the ATA 190 to support fax transmission.

Provides basic testing and troubleshooting

rocedures for the ATA 190.

Provides end-user information about pre-call

mid-call services.

and

nfiguration menu options for the ATA 190.

ications for the ATA

190.

Provides tone parameters for various countries.

• Cisco ATA SIP Compliance Reference Information

http://www-vnt.cisco.com/SPUniv/SIP/documents/CiscoA

• RFC 768 (User Datagram Protocol)

• RFC 2198 (RTP Payload for Redundant Audio Data)

• RFC 2833 (RTP Payload for DTMF Digits, Telephony Phones and Telephony Signals)

• RFC 2327 (SDP: Session Description Protocol)

• RFC 4730 (A Session Initiation Protocol (SIP) Event Package for Key Press Stimulus (KPML))

• RFC 3515 (The Session Initiation Protocol (SIP) Refer Method)

• Read Me First - ATA Boot Load Information

• Cisco ATA 190 Analog Telephone Adapter At a Glance

• Regulatory Compliance and Safety Information for the Cisco ATA 190

• Cisco ATA 190 Analog Telephone Adapter Release Notes

Cisco Unified Communications Manager

TASIPComplianceRef.pdf

These publications are available at the following URL:

http://www.cisco.com/en/US/products/sw/voicesw/ps556/tsd_products_support_series_home.html

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Preface

Cisco Unified Communications Manager Business Edition

These publications are available at the following URL:

http://www.cisco.com/en/US/products/ps7273/

tsd_products_support_series_home.html

Obtaining Documentation, Obtaining Support, and Security Guidelines

For information on obtaining documentation, obtaining support, providing documentation feedback, security guidelines, and also recommended aliases and general Cisco documents, see the monthly What’s New in Cisco Product Documentation, which also lists all new and revised Cisco technical documentation, at:

http://www.cisco.com/en/US/docs/general/whatsnew/whatsnew.html

Cisco Product Security Overview

This product contains cryptographic features and is subject to United States and local country laws governing import, export, transfer and use. Delivery of Cisco cryptographic products does not imply third-party authority to import, export, distribute or use encryption. Importers, exporters, distributors and users are responsible for compliance with U.S. and local country laws. By using this product you agree to comply with applicable laws and regulations. If you are unable to comply with U.S. and local laws, return this product immediately.

Further information regarding U.S. exp

http://www.access.gpo.gov/bis/ear/ear_data.html.

Document Conventions

This document uses the following conventions:

Convention Description

boldface font Commands and keywords are in boldface.

ont Arguments for which you supply values are in italics.

italic f

[ ] Elements in square brackets are optional.

{ x | y | z } Alternative keywords are grouped in braces and separated by vertical bars.

[ x | y | z ] Optional alternative keywords are grouped in brackets and separated by vertical bars.

string A nonquoted set of characters. Do not use quotation marks around the string or the

ing will include the quotation marks.

str

screen font Terminal sessions and information the system displays are in screen font.

boldface

font

screen

italic screen font

Information you must enter is in

Arguments for which you supply values are in it

ort regulations may be found at

boldface screen font.

alic screen font.

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Preface

Convention Description

^ The symbol ^ represents the key labeled Control—for example, the key combination

^D in a screen display means hold down the Control key while you press the D key.

< > Nonprinting characters, such as passw

Note Means reader take note. Notes contain helpful suggestions or references to material not covered in the

publication.

Caution Means reader be careful. In this situation, you might do something that could result in equipment

damage or loss of data.

ords are in angle brackets.

Warning

Means danger. You are in a situation that could cause bodily injury. Before you work on any equipment, be aware of the hazards involved with electrical circuitry and be familiar with standard practices for preventing accidents.

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CHAPTER

Cisco ATA 190 Analog Telephone Adapter Overview

This section describes the hardware and software features of the Cisco ATA 190 Analog Telephone Adapter (ATA 190) and includes a brief overview of the Session Initiation Protocol (SIP).

The ATA 190 analog telephone adapters are handset-to phones to operate on IP-based telephony networks. The ATA 190 supports two voice ports, each with an independent phone number. The ATA 190 also has an RJ-45 10/100BASE-T data port.

This section covers these topics:

• Session Initiation Protocol Overview, page 1-2

• Hardware Overview, page 1-4

• Software Features, page 1-4

• Installation and Configuration Overview, page 1-8

-Ethernet adapters that allow regular analog

Figure 1-1 Cisco Analog Telephone Adapter

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Session Initiation Protocol Overview

Session Initiation Protocol (SIP) is the Internet Engineering Task Force (IETF) standard for real-time calls and conferencing over Internet Protocol (IP). SIP is an ASCII-based, application-layer control protocol (defined in RFC3261) that can be used to establish, maintain, and terminate multimedia sessions or calls between two or more endpoints.

Like other Voice over IP (VoIP) protocols, SIP is designed to address the functions of signaling and

on management within a packet telephony network. Signaling allows call information to be carried

sessi across network boundaries. Session management provides the ability to control the attributes of an end-to-end call.

Note SIP for the ATA 190 is compliant with RFC2543.

This section contains these topics:

• SIP Capabilities, page 1-2

• Components of SIP, page 1-2

Chapter 1 Cisco ATA 190 Analog Telephone Adapter Overview

SIP Capabilities

SIP provides these capabilities:

• Determines the availability of the target endpoint. If a call cannot be completed because the target

• Determines the location of the target endpoint. SIP supports address resolution, name mapping, and

• Determines the media capabilities of the target endpoint. Using the Session Description Protocol

• Establishes a session between the originating and target endpoint. If the call can be completed, SIP

• Handles the transfer and termination of calls. SIP supports the transfer of calls from one endpoint

endpoint is unavailable, SIP determines whether the called party is already on the phone or did not answer in the allotted number of rings. SIP then returns a message indicating why the target endpoint was unavailable.

call redirection.

(SDP), SIP determines the lowest level of common services between endpoints. Conferences are established using only the media capabilities that are supported by all endpoints.

establishes a session between the endpoints. SIP also supports mid-call changes, such as adding another endpoint to the conference or changing the media characteristic or codec.

to another. During a call transfer, SIP establishes a session between the transferee and a new endpoint (specified by the transferring party) and terminates the session between the transferee and the transferring party. At the end of a call, SIP terminates the sessions between all parties. Conferences can consist of two or more users and can be established using multicast or multiple unicast sessions.

Components of SIP

SIP is a peer-to-peer protocol. The peers in a session are called User Agents (UAs). A user agent can function in one of these roles:

• User agent client (UAC)—A client application that initiates the SIP request.

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Chapter 1 Cisco ATA 190 Analog Telephone Adapter Overview

SIP user agents

RTP

SIP

SIP proxy and

redirect servers

SIP gateway

PSTN

Legacy PBX

SIP SIP

72342

• User agent server (UAS)—A server application that contacts the user when a SIP request is received

and returns a response on behalf of the user.

Typically, a SIP endpoint is capable of functioning as both a UAC and a UAS, but functions only as one

r the other per transaction. Whether the endpoint functions as a UAC or a UAS depends on the UA that

o initiated the request.

Session Initiation Protocol Overview

From an architectural standpoint, t

he physical components of a SIP network can also be grouped into

two categories—Clients and servers. Figure 1-2 illustrates the architecture of a SIP network.

Note SIP servers can interact with other application services, such as Lightweight Directory Access Protocol

(LDAP) servers, a database application, or an extensible markup language (XML) application. These application services provide back-end services such as directory, authentication, and billable services.

Figure 1-2 SIP Architecture

SIP Clients

SIP Servers

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SIP clients include:

• Gateways—Provide call control. Gateways provide many services, the most common being a

translation function between SIP conferencing endpoints and other terminal types. This function includes translation between transmission formats and between communications procedures. In addition, the gateway also translates between audio and video codecs and performs call setup and clearing on both the LAN side and the switched-circuit network side.

• Phones—Can act as either a UAS or UAC. The ATA 190 can initiate SIP requests and respond to

requests.

SIP servers include:

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Hardware Overview

• Proxy server—The proxy server is an intermediate device that receives SIP requests from a client

and then forwards the requests on the client’s behalf. Proxy servers receive SIP messages and forward them to the next SIP server in the network. Proxy servers can provide functions such as authentication, authorization, network access control, routing, reliable request retransmission, and security.

• Redirect server—Receives SIP requests, strips out the address in the request, checks its address

tables for any other addresses that may be mapped to the address in the request, and then returns the results of the address mapping to the client. Redirect servers provide the client with information about the next hop or hops that a message should take, then the client contacts the next hop server or UAS directly.

• Registrar server—Processes requests from UACs for registration of their current location. Registrar

servers are often co-located with a redirect or proxy server.

Hardware Overview

The ATA 190 is a compact, easy to install device. Figure 1-3 shows the rear panel of the ATA 190.

Chapter 1 Cisco ATA 190 Analog Telephone Adapter Overview

Figure 1-3 ATA 190—Rear View

390906

The unit provides these connectors and indicators:

• 5V power connector.

• Two RJ-11 FXS (Foreign Exchange Station) ports—The ATA 190 supports two independent RJ-11

phone ports that can connect to any standard analog phone device. Each port supports either voice calls or fax sessions, and both ports can be used simultaneously.

• The ATA 190 has one network port—an RJ-45 10/100BASE-T data port to connect an

Ethernet-capable device, such as a computer, to the network.

Note The ATA 190 performs auto-negotiation for duplexity and speed and is capable of 10/100 Mbps,

full-duplex operation.

Software Features

The ATA 190 supports these protocols, services and methods:

• Secure Real-Time Transport Protocol, page 1-5

• Name Signaling Event based passthrough, page 1-5

• Transport Layer Security Protocol, page 1-5

• T.38 Fax Relay, page 1-5

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Chapter 1 Cisco ATA 190 Analog Telephone Adapter Overview

• Voice Codecs Supported, page 1-5

• Other Supported Protocols, page 1-6

• ATA 190 SIP Services, page 1-6

• Modem Standards, page 1-7

• Fax Services, page 1-7

• Methods Supported, page 1-7

• Supplementary Services, page 1-8

Secure Real-Time Transport Protocol

SRTP (Secure Real-Time Transport Protocol) secures voice conversations on the network and provides protection against replay attacks.

Note Currently ATA190 does not support secure conference call. A 2-way secure call is supported.

Software Features

Name Signaling Event based passthrough

Name Signaling Event (NSE)-based passthrough is simply the transport of fax or modem communications using the G.711 codec.

The ATA 190 does not support NSE-based modem passthrough.

Transport Layer Security Protocol

Transport Layer Security (TLS) is a cryptographic protocol that secures data communications such as e-mail on the Internet. TLS is functionally equivalent to Secure Sockets Layer (SSL).

T.38 Fax Relay

The T.38 fax relay feature enables devices to use fax machines to send files over the IP network. In general, when a fax is received, it is converted to an image, sent to the T.38 fax device, and converted back to an analog fax signal. T.38 fax relays configured with voice gateways decode or demodulate the fax signals before they are transported over IP. With the SIP call control protocol, the T.38 fax relay is indicated by Security Description (SDP) entries in the initial SIP INVITE message. After the initial SIP INVITE message, the call is established to switch from voice mode to T.38 mode. Cisco Unified Communications Administration allows you to configure a SIP profile that supports T.38 fax communication.

Voice Codecs Supported

The ATA 190 supports these voice codecs (check your other network devices for the codecs they support):

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Software Features

• G.711µ-law

• G.711A-law

• G.729A

• G.729AB

Other Supported Protocols

The ATA 190 supports these additional protocols:

• 802.1Q VLAN tagging

• Cisco Discovery Protocol (CDP)

• Domain Name System (DNS)

• Dynamic Host Configuration Protocol (DHCP)

• Internet Control Message Protocol (ICMP)

• Internet Protocol (IP)

• Real-Time Transport Protocol (RTP)

Chapter 1 Cisco ATA 190 Analog Telephone Adapter Overview

• Transmission Control Protocol (TCP)

• Trivial File Transfer Protocol (TFTP)

• User Datagram Protocol (UDP)

ATA 190 SIP Services

These services include these features:

• IP address assignment—DHCP-provided or statically configured

• ATA 190 configuration by Cisco Unified Communications Manager configuration interface

• VLAN configuration

• Cisco Discovery Protocol (CDP)

• Low-bit-rate codec selection

• User authentication

• Configurable tones (dial tone, busy tone, alert tone, reorder tone, call waiting tone)

• Dial plans

• SIP proxy server redundancy

• Privacy features

• User-configurable, call waiting, permanent default setting

1-6

• Comfort noise during silent period when using G.711u/a and G.729ab

• Advanced audio mode

• Caller ID format

• Ring cadence format

• Silence suppression

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Chapter 1 Cisco ATA 190 Analog Telephone Adapter Overview

• Hookflash detection timing configuration

• Configurable onhook delay

• Type of Service (ToS) configuration for audio and signaling ethernet packets

• Debugging and diagnostic tools

Modem Standards

The ATA 190 supports the following modem standards:

• V. 9 0

• V. 9 2

• V. 4 4

• K56Flex

• ITU-T V.34 Annex 12

• ITU-T V.34

• V.32bis

Software Features

Fax Services

Note Success of fax transmission depends on network conditions and fax modem response to these conditions.

• V. 3 2

• V. 2 1

• V. 2 2

• V. 2 3

The ATA 190 supports two modes of fax services, in which fax signals are transmitted using the G.711 codec:

• Fax pass-through mode—Receiver-side Called Station Identification (CED) tone detection with

automatic G.711A-law or G.711µ-law switching.

• T.38 Fax Relay mode: The T.38 fax relay feature enables devices to use fax machines to send files

over the IP network. In general, when a fax is received, it is converted to an image, sent to the T.38 fax device, and converted back to an analog fax signal. T.38 fax relays configured with voice gateways decode or demodulate the fax signals before they are transported over IP.

The network must have reasonably low network jitter, network delay, and packet loss rate.

Methods Supported

The ATA 190 supports these methods. For more information, see RFC3261 (SIP: Session Initiation Protocol).

• REGISTER

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Installation and Configuration Overview

• REFER

• INVITE

• BYE

• CANCEL

• NOTIFY

• OPTIONS

• ACK

• SUBSCRIBE

Supplementary Services

SIP supplementary services are services that you can use to enhance your phone service. For information on how to use these services, see Chapter 6, “Using SIP Supplementary Services”.

The ATA 190 supports these SIP supplementary services:

• Caller ID

• Call-waiting caller ID

Chapter 1 Cisco ATA 190 Analog Telephone Adapter Overview

• Voice mail indication

• Making a conference call

• Call waiting

• Call forwarding

• Calling-line identification

• Unattended transfer (blind transfer)

• Attended transfer

• Shared Line

• SpeedDial

• Conference (MeetMe)

• Pick Up

• Redial

Installation and Configuration Overview

Table 1-1 provides the basic steps required to install and configure the ATA 190 to make it operational

in a typical SIP environment where a large number of ATA 190s must be deployed.

1-8

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Chapter 1 Cisco ATA 190 Analog Telephone Adapter Overview

Ta b l e 1-1 Overview of the Steps Required to Install and Configure the ATA 190 and Make it

Operational

Action Reference

1. Plan the network and ATA 190 configuration.

2. Install the Ethernet connection.

3. Install and configure the other network devices.

4. Install the ATA 190 but do not power up the

ATA 190 yet.

5. Power up the ATA 190.

Installation and Configuration Overview

Installing the ATA 190, page 3-2

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Installation and Configuration Overview

Chapter 1 Cisco ATA 190 Analog Telephone Adapter Overview

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CHAPTER

Preparing to Install the ATA 190 on Your Network

The ATA 190 enables you to communicate using voice over a data network. To provide this capability, the ATA 190 depends upon and interacts with several other key Cisco Unified IP Telephony and network

onents, including Cisco Unified Communications Manager, DNS and DHCP s

comp servers, media resources, and so on.

This chapter focuses on the interactions be DNS and DHCP servers, TFTP servers, and switches. It also describes options for powering the ATA 190.

For related information about voice and IP communications, see this URL:

http://www.cisco.com/en/US/products/sw/voicesw/index.html

This chapter provides an overview of the interaction between the ATA 190 and other key components of

oice over IP (VoIP) network. It includes these topics:

the V

• Understanding Interactions with Other Cisco Unified IP Communications Products, page 2-1

• Providing Power to the ATA 190, page 2-2

• Understanding Phone Configuration Files, page 2-3

• Understanding the ATA 190 Startup Process, page 2-4

tween the ATA 190, Cisco Unified Communications Manager,

ervers, TFTP

• Adding the ATA 190 to the Cisco Unified Communications Manager Database, page 2-5

• Determining the MAC Address of an ATA 190, page 2-7

Understanding Interactions with Other Cisco Unified IP Communications Products

To function in the IP telephony network, the ATA 190 must be connected to a networking device, such as a Cisco Catalyst switch. You must also register the ATA 190 with a Cisco Unified Communications

nager system before sending and receiving calls.

This section includes information on U

Unified Communications Manager, page 2-2.

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nderstanding How the ATA 190 Interacts with Cisco

2-1

Page 22

Chapter 2 Preparing to Install the ATA 190 on Your Network

Providing Power to the ATA 190

Understanding How the ATA 190 Interacts with Cisco Unified Communications Manager

Cisco Unified Communications Manager is an open and industry-standard call processing system. Cisco Unified Communications Manager software sets up and tears down calls between phones

nected to the ATA 190, integrating traditional PBX functionality with the corporate IP network.

con Cisco Unified Communications Manager manages the component phones, the access gateways, and the resources necessary for features such as call conferencing and route planning. Cisco Unified Communications Manager also provides:

• Firmware for devices

• Authentication and encryption (if configured for the telephony system)

• Configuration and CTL files via the TFTP service

• Phone registration

• Call preservation, so that a media session continues if signaling is lost between the primary

Communications Manager and a phone

For information about configuring Cisco Unified Communications Manager to work with the IP devices

scribed in this chapter, see Cisco Unified Communications Manager Administration Guide, Cisco

nified Communications Manager System Guide, and Cisco Unified Communications Manager Security

U Guide.

s of the IP telephony system—the

Providing Power to the ATA 190

The ATA 190 is powered with external power. External power is provided through a separate power supply.

The following sections provide more information about powering a ATA 190:

• Power Guidelines, page 2-2

• Power Outage, page 2-2

• Understanding Phone Configuration Files, page 2-3

Power Guidelines

The following power type and guideline applies to external power for the ATA 190:

• Power Type—External power (Provided through the Universal AC external power supply.)

• Guidelines—The ATA 190 uses the Universal AC power supply 110/240V

Power Outage

Your accessibility to emergency service through the phone is dependent on the phone being powered. If there is an interruption in the power supply, Service and Emergency Calling Service dialing will not function until power is restored. In the case of a power failure or disruption, you may need to reset or reconfigure equipment before using the Service or Emergency Calling Service dialing.

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Chapter 2 Preparing to Install the ATA 190 on Your Network

Understanding Phone Configuration Files

Configuration files for a phone are stored on the TFTP server and define parameters for connecting to Cisco Unified Communications Manager. In general, any time you make a change in Cisco

ied Communications Manager that requires the phone to be reset, a ch

Unif to the phone’s configuration file. If the system needs to reset or restart, both ports must reset or restart at the same time.

Configuration files also contain information about which imag image load differs from the one that is currently loaded on a phone, the phone contacts the TFTP server to request the required load files. (These files are digitally signed to ensure the authenticity of the file source.)

In addition, if the device security mode in the configurat on the phone has a valid certificate for Cisco Unified Communications Manager, th a TLS connection to Cisco Unified Communications Manager. Otherwi TCP/UDP connection. For SIP phones, a TLS connection requires that the transport protocol in the phone configuration file be set to TLS, which corresponds to the transport type in the SIP Security Profile in Cisco Unified Communications Manager.

ion file is set to Authenticated and the CTL file

Understanding Phone Configuration Files

ange is automatically made

e load the phone should be running. If this

e phone establishes

se, the phone establishes a

If you configure security-related settings in Cisco Unif

ied Communications Manager Administration, the phone configuration file will contain sensitive information. To ensure the privacy of a configuration file, you must configure it for encryption. For detailed information, see Configuring Encrypted Phone

Configuration Files in Ci

A phone accesses a default configuratio

sco Unified Communications Manager Security Guide.

n file named XMLDefault.cnf.xml only when the phone has not received a valid Trust List file containing a certificate assigned to the Cisco Unified Communications Manager and TFTP.

If auto registration is not enabled and you did not add the phone to the Cisco Unified Communications

ager database, the phone does not attempt to register with Cisco Unified Communications Manager.

Man

If the phone has registered before, the phone accesses th

e configuration file named

ATA <mac_address>.cnf.xml, where mac_address is the MAC address of the phone.

• Configuration Files:

–

For unsigned and unencrypted files—ATA<mac>.cnf.xml

–

For signed files—ATA<mac>.cnf.xml.sgn

–

For signed and encrypted files—ATA<mac>.cnf.xml.enc.sgn

• Dial Plan—<dialplan>.xml

–

Support “,” for second dial tone

–

No support > for configuring termination key

–

No support + dial pattern which contains + will be ignored

–

Maximum number of dial pattern is 10

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–

Maximum length of each dial pattern is 30

The filenames are derived from the MAC Address and Description fields in the Phone Configuration

indow of Cisco Unified Communications Manager Administration. The MAC address uniquely

entifies the phone. For more information see the Cisco Unified Communications Manager

id Administration Guide.

Cisco ATA 190 Analog Telephone Adaptor Administration Guide for SIP (Version 1.0)

2-3

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Chapter 2 Preparing to Install the ATA 190 on Your Network

Understanding the ATA 190 Startup Process

For more information about how the phone interacts with the TFTP server, see the Cisco Unified

Communications Manager System Guide, Cisco TFTP section.

Understanding the ATA 190 Startup Process

When connecting to the VoIP network, the ATA 190 goes through a standard startup process, as described in Table 2-1. Depending on your specific network configuratio on your ATA 190.

Ta b l e 2-1 ATA 190 Startup Process

Task Purpose Related Topics

1. Obtaining Power.

The ATA 190 uses external power.

2. Loading the Stored Image.

The ATA 190 has non-volatile flash memory in which it

tores firmware images and user-defined preferences. At

s startup, the phone runs a bootstrap loader that loads a phone image stored in flash memory. Using this image, the phone initializes its software and hardware.

3. Obtaining an IP Address.

See Providing Power to the ATA 190, page 2-2.

n, not all of these process steps may occur

If the ATA 190 is using DHCP to obtain an IP address, the

vice queries the DHCP server to obtain one. If you are

de not using DHCP in your network, you must assign static IP addresses to each device locally.

4. Requesting the CTL file.

The TFTP server stores the CTL file. This file contains the

ificates necessary for establishing a secure connection

cert between the device and Cisco Unified Communications Manager

See the C

Security Guide, Configuring the Cisco CTL

Client.

isco Unified Communications Manager

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Chapter 2 Preparing to Install the ATA 190 on Your Network

Adding the ATA 190 to the Cisco Unified Communications Manager Database

Table 2-1 ATA 190 Startup Process (continued)

Task Purpose Related Topics

5. Requesting the Configuration File.

The TFTP server has configuration files, which define paramet Communications Manager and other information for the ATA 190.

6. Contacting Cisco Unified Communications Manager.

The configuration file defines how the ATA 190 co Manager and obtaining the file from the TFTP server, the device attempts to make a connection to the highest priority Cisco Unified Communications Manager on the list. If the securi signaling (encrypted or authenticated), and the Cisco Unified Communications Manager is set to secure mode, the device makes a TLS connection. Otherwise, it makes a nonsecure TCP/UDP connection.

ers for connecting to Cisco Unified

mmunicates with Cisco Unified Communications

provides a device with its load ID. After

ty profile of the device is configured for secure

nderstanding Phone Configuration Files,

See U

page 2-3.

nderstanding Phone Configuration Files,

See U

page 2-3.

Adding the ATA 190 to the Cisco Unified Communications Manager Database

Before installing the ATA 190, you must choose a method for adding the devices to the Cisco Unified Communications Manager database. These sect

• Adding the ATA 190 with Auto-Registration, page 2-6

• Adding the ATA 190 with Cisco Unified Communications Manager Administration, page 2-6

Table 2-2

Cisco Unified Communications Manager database.

Ta b l e 2-2 Methods for Adding the ATA 190 to the Cisco Unified Communications Manager

Method

Auto-registration No

Using the Cisco Unified

ommunications

C Manager Administration

provides an overview of these methods for adding the ATA 190 to the

Database

Requires MAC Address? Notes

• Results in automatic assignment of directory

numbers.

• Not available when mixed mode is enabled.

Yes Requires phones to be added individually.

ions describe the methods:

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Adding the ATA 190 to the Cisco Unified Communications Manager Database

Adding the ATA 190 with Auto-Registration

By enabling auto-registration before you begin installing the ATA 190, you can:

• Add devices without first gathering MAC addresses from the ATA 190.

• Automatically add a ATA 190 to the Cisco Unified Communications Manager database when you

physically connect the phone to your IP telephony network. During auto-registration, Cisco Unified Commu

• Quickly enter devices into the Cisco Unified Communications Manager database and modify any

settings, such as the directory numbers, from Cisco Unified Communications Manager.

• Move auto-registered devices to new locations and assign them to different device pools without

affecting their directory numbers.

Auto-registration is disabled by default. In some cases, you example, if you want to assign a specific directory number to the phone or if you plan to use secure connection with Cisco Unified Communications Manager as described in Cisco Unified Communications Manager Security Guide. For information about enabling auto-registration, see the Enabling Auto-Registration in the Cisco Unified Communications Manager Administration Guide.

nications Manager assigns the next available sequential directory number to the phone.

Chapter 2 Preparing to Install the ATA 190 on Your Network

may not want to use auto-registration; for

Note When you configure the cluster for mixed mode through the Cisco CTL client, auto-registration is

automatically disabled. When you configure the cluster for nonsecure mode through the Cisco CTL client, auto-registration is not automatically enabled.

Related Topics

Adding the ATA 190 with Auto-Registration, page 2-6

Determining the MAC Address of an ATA 190

Several of the procedures that are described in this manual require you to determine the MAC address of an ATA 190. You can determine the MAC address for a device in any of these ways:

• Look at the MAC label on the back of the device.

• Display the web page for the device and click the Device Information hyperlink.

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Determining the MAC Address of an ATA 190

Chapter 2 Preparing to Install the ATA 190 on Your Network

2-8

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Installing the ATA 190

This section describes how to connect the ATA 190 hardware and configure the ATA 190 by loading the QED and firmware files. You must install the QED file first and then install the firmware file.

Cisco ATA Rear Panel Connections

Figure 3-1 Cisco ATA Rear Panel

CHAPTER

PHONE 1/PHONE 2—Connection to Analog telephones or fax.

NETWORK—Co

POWER—Co

nnection to IP network.

nnection to 5V power adapter.

Network Requirements

The ATA 190 acts as an endpoint on an IP telephony network. The following equipment is required:

• Call Control system

• Voice packet gateway—Required if you are connecting to the Public Switched Telephone Network

(PSTN). A gateway is not required if an analog key system is in effect.

• Ethernet connection

Safety Recommendations

To ensure general safety, follow these guidelines:

• Do not get this product wet or pour liquids into this device.

390906

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Page 30

What the ATA 190 Package Includes

• Do not open or disassemble this product.

• Do not perform any action that creates a potential hazard to people or makes the equipment unsafe.

• Use only the power supply that comes with the ATA 190.

Chapter 3 Installing the ATA 190

Warning

Ultimate disposal of this product should be handled according to all national laws and regulations.

Read the installation instructions before you connect the system to its power source.

The plug-socket combination must be accessible at all times because it serves as the main disconnecting device.

Do not work on the system or connect or disconnect cables during periods of lightning activity.

For translated warnings, see the Regulatory Compliance and Safety Information for the Cisco ATA 190 manual.

What the ATA 190 Package Includes

The ATA 190 package contains the following items:

• ATA190 device

• Pointer Card

• 5V Power Adapter with appropriate Country Clip

• Ethernet Cable

Note The ATA 190 is intended for use only with the 5V DC power adapter that comes with the unit.

Installing the ATA 190

To install an ATA 190, follow these steps:

Procedure

Step 1 Connect the power supply to the Cisco DC Adapter port.

Step 2 Connect a straight-through Ethernet cable from the network to the 10/100 SW port on the ATA 190. Each

ATA 190 ships with one Ethernet cable in the box.

Cisco ATA 190 Analog Telephone Adaptor Administration Guide for SIP (Version 1.0)

3-2

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Chapter 3 Installing the ATA 190

Note You can use either Category 3/5/5e/6 cabling for 10 Mbps connections, but you must use Category 5/5e/6

for 100 Mbps connections.

Attaching a Phone to the ATA 190

You can attach one or two phones to an ATA 190 by connecting them to a line port of the ATA 190 with a RJ11 cable. The line LED will blink when there is activity on that line.

Verifying the ATA 190 Startup Process

After the ATA 190 has power connected to it, the phone begins its startup process by cycling through these steps:

1. The Power LED is on.

2. The Network LED is flashing (when there is data traffic on Network port connected to a WAN port.)

The ATA 190 is launching its application.

Attaching a Phone to the ATA 190

3. Network LED is on.

4. After the Phone1 and Phone2 resgister with CUCM successfully, the corresponding LEDs are on.

5. All of the LEDs are on.

If the ATA 190 flash memory is erased it can restore the image by manual upgrading.

When you go offhook on the phone, you will see the line LED to begin flashing, and you will hear dial

one. The ATA 190 has completed the startup process.

or the load is corrupted, the ATA enters a recovery mode where

Configuring Startup Network Settings

It is recommended to use DHCP instead of Static IP. DHCP server provides ip, mask, gateway, tftp server, etc.

Configuring Security on the ATA 190

The security features protect against several threats, including threats to the identity of the phone and to data. These features establish and maintain authenticated communication streams between the phone and the Cisco Unified Communications Manager server, and digitally sign files before they are delivered.

For more information about the sec Security Guide.

urity features, see the Cisco Unified Communications Manager

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You can initiate the installation of a Locally Signi Configuration menu on the phone. This menu also lets you update or remove an LSC.

Before you begin, make sure that the appropriate Cisco Unified Communications Manager and the CAPF

urity configurations are complete:

sec

Cisco ATA 190 Analog Telephone Adaptor Administration Guide for SIP (Version 1.0)

ficant Certificate (LSC) from the Security

3-3

Page 32

Configuring Security on the ATA 190

• On Cisco Unified Communications Operating System Administration, verify that the CAPF

certificate has been installed

• The CAPF is running and configured

Chapter 3 Installing the ATA 190

See the Cisco Un

Note If you want to update LSC, you need to use reset to factory default from Chapter B, “Voice Menu

ified Communications Manager Security Guide for more information.

Codes”.

3-4

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CHAPTER

Configuring the ATA 190

Yo u m us t u se C is co Unified Communications Manager Administration to configure telephony features and assign users.

This chapter provides an overview of these configuration and setup procedures. Cisco Unified Commu

Telephony Features Available for the ATA 190

Table 4-1 lists the supported telephony features, many of which you configure by using

Cisco Unified Communications Manager Administration.

Ta b l e 4-1 Telephony Features for the ATA 190

Feature Description Configuration Reference

Audible Message

aiting Indicator

cBarge Allows a user to join a non-pri

A stutter tone from the handset, headset, or speakerphone indicates that a user has one or more new voice messages on a line.

Note The stutter tone is line-specific. You hear it only

line. cBarge adds a user to a call and converts it into a conference, allowing the user and other parties to access conference features.

The phones support Barge on a shared confere

nications Manager documentation provides detailed instructions for these procedures.

For more information, refer to:

• Cisco Unified Communications

Manager Administration Guide,

Message Waiting Configuration.

when using the line with the waiting messages.

vate call on a shared phone

nce bridge.

• Cisco Unified Communications

Manager System Guide, Voice Mail

Connectivity to Cisco Unified Communications Manager.

For more information, refer to:

• Cisco Unified Communications

Manager Administration Guide,

Cisco Unified IP Phone Configuration.

• Cisco Unified Communications

Manager System Guide, Cisco

Unified IP Phones.

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• Cisco Unified Communications

Manager Features and Services Guide, Barge and Privacy.

Cisco ATA 190 Analog Telephone Adaptor Administration Guide for SIP (Version 1.0)

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Chapter 4 Configuring the ATA 190

Telephony Features Available for the ATA 190

Table 4-1 Telephony Features for the ATA 190 (continued)

Feature Description Configuration Reference

Call forward Allows users to redirect incoming calls to another number.

Call forward options include Call Forward All, Call Forward Busy, and Call Forward No Answer.

For more information, refer to:

• Cisco Unified Communications

Manager Administration Guide,

Directory Number Configuration.

• Cisco Unified Communications

Manager System Guide, Cisco

Unified IP Phones.

Call pickup Allows users to redirect a call that is ringing on another

one within their pickup group to their phone.

You can configure an audio and/or visual alert for the

mary line on the phone. This alert notifies the users that

pri

For more information, refer to the Cisco

fied Communications Manager

Uni Features and Services Guide, Call

Pickup.

a call is ringing in their pickup group.

Call waiting Indicates (and allows users to

answer) an incoming call that rings while on another call. Displays incoming call information on the phone screen.

Caller ID Displays caller identification such as a phone number,

name, or o

ther descriptive text on the phone screen.

For more information, refer to the Cisco Unified Communications System Guide,

Understanding Directory Numbers.

For more information, refer to:

• Cisco Unified Communications

Manager Administration Guide,

Cisco Unified IP Phone Configurations.

• Cisco Unified Communications

Manager System Guide,

Understanding Route Plans.

• Cisco Unified Communications

Manager Features and Services Guide, Call Display Restrictions.

Conference

4-2

• Allows a user to talk simultaneously with multiple

parties by calling each participant individually. Conference features include Adhoc Conference, cBarge, and Meet–Me.

• Allows a non-initiator in a standard (ad hoc) conference

to add or remove participants.

Cisco ATA 190 Analog Telephone Adaptor Administration Guide for SIP (Version 1.0)

• Cisco Unified Communications

Manager Administration Guide,

Directory Number Configuration.

For more information, refer to:

• Cisco Unified Communications

Manager System Guide,

Cisco Unified IP Phones.

• The service parameter, Advance

Adhoc Conference, (disabled by default in Cisco Unified Communications Manager Administration) allows you to enable these features.

Note Be sure to inform your users

whether these features are activated.

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Chapter 4 Configuring the ATA 190

Telephony Features Available for the ATA 190

Table 4-1 Telephony Features for the ATA 190 (continued)

Feature Description Configuration Reference

Direct transfer Allows users to connect two calls to each other (without

remaining on the line).

For more information, refer to the Cisco

Unified Communications Manager System Guide, Cisco Unified IP Phones.

Forced authorization codes (F

AC)

Controls the types of calls that certain users can place.

Note If you are using this feature, you must disable

Enbloc dialing.

For more information, refer to the Cisco

nified Communications Manager

eatures and Services Guide, Client

Matter Codes and Forced Authorization Codes.

Group call pickup Allows a user to answer a call that is ringing on a directory

mber in another group.

For more information, refer to the Cisco

Unified Communications Manager Features and Services Guide, Call

Pickup.

Hold/Resume Allows the user to move a connec

state and a held state.

Note No support for resuming a call from a shared line

party.

Meet–Me

onference

Allows a user to host a Meet-Me conference in which other participants call a predetermined number at a scheduled time.

ted call between an active

For more information, refer to:

Requires no configuration, unless you

ant to use music on hold. See Music on

hold in this table for information.

For more information refer to Cisco

Unified Communications Manager Administration Guide, Meet-Me

Number/Pattern Configuration.

Message Waiting Defines directory numbers for message-waiting on and

message-w

aiting off indicator. A directly connected voice-messaging system uses the specified directory number to set or to clear a message-waiting indication for a particular Cisco Unified IP Phone.

For more information refer to:

• Cisco Unified Communications

Manager Administration Guide,

Message Waiting Configuration.

• Cisco Unified Communications

Manager System Guide, Voi c e M ai l

Connectivity to Cisco Unified Communications Manager.

Music on hold Plays music while cal

lers are on hold. For more information refer to Cisco

Unified Communications Manager Features and Services Guide, Music On

Hold.

Privacy Prevents users who share a line from adding themselves to

call and from viewing information on their phone screens

a about the call of the other user.

For more information refer to:

• Cisco Unified Communications

Manager Administration Guide,

Cisco Unified IP Phone Configuration.

• Cisco Unified Communications

Manager System Guide, Cisco

Unified IP Phones.

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• Cisco Unified Communications

Manager Features and Services Guide Barge and Privacy.

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Chapter 4 Configuring the ATA 190

Configuring Product Specific Configuration Parameters

Table 4-1 Telephony Features for the ATA 190 (continued)

Feature Description Configuration Reference

Redial Allows users to call the most recently dialed phone number

y pressing feature code *#.

Shared line Allows a user to have several phones that share the same

one number or allows a user to share a phone number with

a coworker.

Speed dialing Allows users to speed dial a phone number by entering an

gned index code (1 to 199) on the phone keypad.

assi

Note You can use Speed Dialing while on-hook or

off-hook.

Users assign index codes from the User Options web pages.

Time Zone Update Updates the IP phone with time zone changes. For more information, refer to the Cisco

Voice-messaging system

Enables callers to leave messages if calls are unanswered. For more information refer to:

Requires no configuration.

For more information refer to the Cisco

Unified Communications Manager System Guide, Understanding Directory

Numbers.

For more information, refer to:

• Cisco Unified Communications

Manager Administration Guide,

Cisco Unified IP Phone Configuration.

• Cisco Unified Communications

Manager System Guide, Cisco

Unified IP Phones.

fied Communications Manager

Uni Administration Guide, Date/Time Group

Configuration.

• Cisco Unified Communications

Manager Administration Guide,

Cisco Voice-Mail Port Configuration.

• Cisco Unified Communications

Manager System Guide, Voi c e M ai l

Connectivity to Cisco Unified Communications Manager.

Configuring Product Specific Configuration Parameters

Cisco Unified Communications Manager Administration allows you to set some product specific configuration parameters for the ATA 190. Table 4-2 lists the configuration windows and their paths to configure the parameters.

Ta b l e 4-2 Configuration Information

Configuration Window Path

Enterprise Phone

nfiguration window

Common Phone Profile wi

ndow

Phone Configuration window Device > Phone;

System > Enterprise Phone Configuration

Device > Device Settings > Common Phone Profile

Product Specific Configuration portion of window

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Chapter 4 Configuring the ATA 190

Table 4-3 lists the configuration parameters you can set using Cisco Unified Communications Manager

Administration.You can set the configu listed in Table 4-2.

When you set the parameters, select the to update. If you do not check this box, the corresponding parameter setting does not take effect.

Note Some ATA 190 parameters are set from port 1 only. Setting the parameters from port 2 will have no

effect. Set the following parameters from port 1 only—IVR Password, Phone Load Name, CDP, and Web Access.

Ta b l e 4-3 Configuration Parameters for the ATA 190

Parameter Description

Auto Barge Auto Barge adds a user to an active call. An offhook phone

Call Sequence

Configuring Product Specific Configuration Parameters

ration parameters using any of the three configuration windows

Override Common Settings check box for each setting you wish

automatic

ally adds the user (initiator) to the shared line call (target), and the users currently on the call receive a tone (if configured). Barge supports built-in conference and shared conference bridges.

The Auto Barge feature allows the user to go offhook and be

d to the call. The Auto Barge feature supports built-in

adde conferences and shared conference bridges.

• Bellcore FSK

• ETSI FSK

Caller ID

• BT FSK

• Bellcore FSK

• ETSI FSK

Cisco Discovery Protocol (CDP) Enable or disable the CDP function of the ATA 190

Fax Error Correction Mode Override You can set the fax error correction mode override values to

ne of the following settings:

• Default

• On

• Off

Fax Mode The Cisco ATA supports two fax modes:

• Fax Pass–Through—Allows fax and modem traffic to

pass through a voice port

• T.38 Fax Relay—Allows for a more robust protocol for

fax transmission over packet networks

• NSE Fax pass–Through—g711ulaw

• NSE Fax pass–Through—g711alaw

Hookflash Timer

(100 ms to 1500 ms)

Hookflash Timer

The time to validate hookflash event.

Impedance The ATA 190 provides multiple impedance values, such as

0 ohm for use in the United States

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Adding Users to Cisco Unified Communications Manager

Table 4-3 Configuration Parameters for the ATA 190 (continued)

Parameter Description

Input Audio Level Gain value of Network–to–Phone

IVR Password ATA 190 IVR password

Offhook Validation Timer

Chapter 4 Configuring the ATA 190

Offhook Validation Timer

(50 ms to 1000 ms)

Onhook Delay Timer

(0 s to 155 s)

Onhook Validation Timer Onhook Validation Timer

Output Audio Level Gain value of Phone–to–Network

Indicates the time to validate an offhook event

On-hook Delay Timer

Indicates the time to delay an onhook event

Note This parameter is reserved but does take effect now.

Indicates time to validate an onhook event

Adding Users to Cisco Unified Communications Manager

Adding users to Cisco Unified Communications Manager allows you to display and maintain information about users and allows each user to perform these tasks:

• Access the corporate directory and other customized directories from an ATA 190.

• Create a personal directory.

• Set up speed dial and call forwarding numbers.

• Subscribe to services that are accessible from an ATA 190.

You can add users to Cisco Unified Communications Manager using either of these methods:

• To add users individually, choose User Management > End User from

Cisco Unified Communications Manager Administration.

4-6

Refer to C adding users. Refer to Cisco Unified Communications Manager System Guide for details about user information.

Cisco ATA 190 Analog Telephone Adaptor Administration Guide for SIP (Version 1.0)

isco Unified Communications Manager Administration Guide for more information about

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Configuring Fax Services

The ATA 190 provides two modes of fax services that are capable of internetworking with Cisco IOS gateways over IP networks. These modes are called fax pass-through mode and T.38 fax relay mode.

With fa it through the Voice Over IP (VoIP) network as though the fax were a voice call.

With T standard fax terminals communicating over SIP networks. T.38 fax relay mode provides a more reliable and error-free method of sending faxes over an IP network

Using Fax Mode

You can choose the preferred fax mode on the phone configuration page of the Unified CM administration page. From the fax mode pull-down window, choose one of the following modes:

• Fax pass-through

• T.38 fax r e lay

x pass-through mode, the ATA 190 encodes fax traffic within the G.711 voice codec and passes

.38 fax relay mode, the ATA 190 supports the transmission of faxes, in real time, between two

CHAPTER

• NSE Fax pass–Through—g711ulaw

• NSE Fax pass–Through—g711alaw

You can set the Fax Error correction mode override values. From the fax mode pull-down window,

ose one of the following modes:

cho

• On

• Off

• Default

Fax Modem Standards

Note V.34 is supported for fax.

The ATA 190 supports the following fax modem standards:

• ITU-T V.34

• ITU-T V.34 Annex 12

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Using Fax Mode

• K56flex

• V. 2 1

• V. 2 2

• V. 2 3

• V. 3 2

• V.32bis

• V. 4 4

• V. 9 0

• V. 9 2

Fax Modem Speeds

The ATA 190 supports the following fax modem speeds:

• 14.4 kb/s

• 12 kb/s

Chapter 5 Configuring Fax Services

• 9.6 kb/s

• 7.2 kb/s

• 4.8 kb/s

• 2.4 kb/s

5-2

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Using SIP Supplementary Services

SIP supplementary services are services that you can use to en hance your telephone service. These services include call forward, redial, call forwarding, and conference calling.

Common Supplementary Services

The supplementary services described in this section, and their configuration and implementation, depend on the system of the country in which the service is activated. For information about your country’s implementation of services, contact your local Cisco equipment provider.

This section contains the fo

• Attended Transfer, page 6-2

• Call Pickup, page 6-2

• Caller ID, page 6-2

llowing topics:

CHAPTER

• Call-Waiting Caller ID, page 6-2

• Call Hold, page 6-2

• Group Call Pickup, page 6-3

• Meet–Me Conference, page 6-3

• Privacy, page 6-3

• Shared Line, page 6-3

• Speed Dial, page 6-4

• Redial, page 6-4

• Unattended Transfer, page 6-4

• Voice Mail Indication, page 6-5

• Voice-Messaging System, page 6-5

• Making a Conference Call in the United States, page 6-5

• Making a Conference Call in Sweden, page 6-5

• Call Waiting in the United States, page 6-6

• Call Waiting in Sweden, page 6-6

• About Call Forwarding, page 6-6

• Call Forwarding in the United States, page 6-6

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Common Supplementary Services

• Call Forwarding in Sweden, page 6-6

Attended Transfer

This feature allows a user to tr ansfer an exist ing ca ll to anot her phone number after first consultin g with the dialed party before the user hangs up. Perform the following steps to complete an attended transfer:

Procedure

Step 1 Press the flash button on the phone handset to put the existing party on hold and get a dial tone. Step 2 Dial the phone number to which the existing party is being transferred. Step 3 When the callee answers the phone, you ma y c o n su l t w i t h t h e c a l le e and then transfer the existing party

by hanging up your phone handset.

Chapter 6 Using SIP Supplementary Services

Call Pickup

Allows you to answer a call that is ringing on an oth er ph one withi n your call pickup grou p. Perform th e following steps to use the call pickup feature:

Procedure

Step 1 Pick up the phone handset. Step 2 Press **3.

Caller ID

When the phone rings, the ATA 190 sends a Caller ID signal to the phone between the first and second ring (with name, phone number, time, and date information, if these are available).

Call-Waiting Caller ID

The A TA 190 plays a call waiting tone, then sends an off-hook Caller ID signal to the phone immediately after the first tone burst.

The ATA 190 sends the name, phone number, time, and date information, if these are available.

Call Hold

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This feature allows the user to place an active state in a held state.

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Chapter 6 Using SIP Supplementary Services

Group Call Pickup

Allows you to answer a call on a phone that is outside your call pickup group by:

• Using a group pickup number

• Dialing the ringing phone's number

Perform the following steps to use the group call pickup feature:

Procedure

Step 1 Pick up the phone handset. Step 2 Press **4 > group ID > #.

Meet–Me Conference

Common Supplementary Services

Privacy

This feature allows a user to host a Meet–Me conference in which other participants call a predetermined number at a scheduled time. Perform the following steps to complete a meet–me conference:

Procedure

Step 1 Pick up the phone handset. Step 2 Press **5 > roo m ID > #.

This feature prevents users who share a line from adding themselves to a call and from viewing information on their phone screens about the call of the other user. Perform the following steps to enable or disable the privacy feature:

Procedure

Step 1 Pick up the phone handset. Step 2 During an active call, press **8 to enable the privacy feature. Step 3 During an active call, press **9 to disable the privacy feature.

Shared Line

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This feature allows a user to have multiple phones that share the same phone number or allows a user to share a phone number with a coworker. It enables the phone lines to barge into an existing call.

—If auto barge is enabled, off hook triggers C-barge.

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Common Supplementary Services

—If auto barge is disabled, pressing “**6” triggers C-barge.

Speed Dial

This feature allows users to speed dial a phone number by entering an assigned index code (1 to 199) on the phone keypad.

Redial

Allows users to call the most recently dialed phone number by pressing the *# buttons.

Unattended Transfer

This feature allows a user to transfer an existing call to another phone number without waiting for the dialed party to answer before the user hangs up. Two methods exist for performing an unattended transfer:

Chapter 6 Using SIP Supplementary Services

• Semi-unattended Transfer, page 6-4

• Fully Unattended Transfer (Blind Transfer), page 6-4

Semi-unattended Transfer

Perform the following steps to complete a semi-unattended transfer:

Procedure

Step 1 Press the flash button on the phone handset to put the other party on hold and get a dial tone. Step 2 Dial the phone number to which you would like to transfer the other party. Step 3 Wait for at least one ring and then ha ng up your phone to tran sfer the other part y.

Fully Unattended Transfer (Blind Transfer)

Perform the following steps to complete a fully unattended transfer:

Procedure

Step 1 Press the flash button on the phone handset to put the other party on hold and get a dial tone. Step 2 Press #90 (the transfer service activation code) on your phone keypad, then enter the phone number to

which you want to transfer the other pa rty, then press #.

Step 3 Hang up your phone.

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Chapter 6 Using SIP Supplementary Services

Voice Mail Indication

This feature allows the ATA 190 to play an intermitt ent d ial tone if there is a message in the u ser's voice mail box.

Voice-Messaging System

This feature enables callers to leave messages if calls are unanswered or access voice messages. Perform the following steps to access the voice-messaging system:

Procedure

Step 1 Pick up the phone handset. Step 2 Press *0.

Common Supplementary Services

Making a Conference Call in the United States

Procedure

Step 1 Dial the first number. Step 2 When the person you called answers, press the flash or receiver button on the phone handset. This will

put the first person you called on hold and you will receive a dial tone.

Step 3 Dial the second person and speak normally when that person answers. Step 4 To conference with both callers at the same time, perform a hook flash.

Making a Conference Call in Sweden

Procedure

Step 1 Dial the first number. Step 2 When the person you called answers, press the flash or receiver button on the phone handset. This will

put the first person you called on hold and a dial tone will sound.

Step 3 Dial the second person and speak normally when that person answers. Step 4 Perform a hook flash, then press 2 on your phone keypad to return to the first person. You can continue

to switch back and forth between the two callers.

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Step 5 To conference with both callers at the same time, perform a hook flash, t hen press 3 on the phone keypad.

Once you conference all three callers, the only way to drop a caller is for that caller to hang up.

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Common Supplementary Services

Call Waiting in the United States

If someone calls you while you are speaking on the phone, you can answer by performing a hook flash but cannot conference in all three callers.

Call Waiting in Sweden

If someone calls you while you are speaking on the phone, you can answer by performing a hook flash then pressing 2 on your phone keypad, or you can conference them with the person to whom you are already speaking by performing a hook flash then pressing 3. You can also perform a hook flash then press 3 later during the call to create a conference call.

Chapter 6 Using SIP Supplementary Services

Performing a hook flash then pressing 1 hangs up no answer after one minute, the caller receives three beeps and a busy signal.

T o enable call waiting for Sweden, go to ETSI FSK for Sweden.

Note In ETSI mode, the user must pick up the call waiting rather than start the conference service. The user

cannot trigger the call conference service directly, when there is a call waiting.

to A TA190's configuration webpage and change the Call Sequence

About Call Forwarding

In SIP, the ATA 190 can control call forwarding and call return.The type of call forwarding that is supported for the ATA 190 is Forward Unconditional—Forwards every call that comes in.

Call Forwarding in the United States

Forward Unconditional

Press #72 on your phone keypad; enter the number you want to forward call to; then press # again.

Cancelling Call Forwarding

To cancel call forwarding, press #73 on your phone keypad

the first caller and answers the second call. If there is

Call Forwarding in Sweden

Forward Unconditional

Press *21* on your phone keypad; enter the number you want to forward calls to; then press #. To cancel, press #21#.

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ATA 190 Specifications

This section describes ATA190 specifications:

• Physical Specifications, page A-1

• Electrical Specifications, page A-2

• Environmental Specifications, page A-2

• Physical Interfaces, page A-3

• Ringing Characteristics, page A-3

• Software Specifications, page A-3

• SIP Compliance Reference Information, page A-4

Physical Specifications

APPENDIX

Ta b l e A-1 Physical Specifications

Description Specification

Regulatory compliance FCC (Part 15 Class B), CE, ICES-003, A-Tick

fication, Restriction of Hazardous Substances

certi (RoHS), and UL

Power supply DC input voltage: 5V DC at 2.0A maximum power

sumption: 5W

con

Switching type (100-240V): Automatic

Power adapter: 100-240V and 50-60 Hz (26-34 VA) AC

nput with 1.8m cord

Indicator lights and LEDs Phone 1, phone 2, internet, and power

Documentation Quick Start Guide

Administration Guide (available online)

Provisioning Guide (available online)

Environmental

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Electrical Specifications

Table A-1 Physical Specifications

Description Specification

Dimensions (W x H x D) 3.98 x 3.98 x 1.10 in. (101 x 101 x 28 mm)

Unit weight 5.40 oz (153 g)

Operating temperature 32 to 113ºF (0 to 45ºC)

Storage temperature -77 to 158ºF (-25 to 70ºC)

Operating humidity 10% to 90% noncondensing

Storage humidity 10% to 90% noncondensing

Electrical Specifications

Ta b l e A-2 Electrical Specifications

Description Specification

Power 0.25 to 12W (idle to peak)

DC input voltage 5.0 V at 2.0A maximum

Power adapter Universal AC/DC

~4.05 x 1.93 x 1.31 in. (~10.3 x 4.9 x 3.35 cm)

Appendix A ATA 190 Specifications

~4.23 oz (120 g) for the AC-input external power adapter

~4.9 ft (1.5 m) DC cord

6 ft (1.8 m) cord

UL/cUL, CE approved

Class I adapter

Environmental Specifications

Ta b l e A-3 Environmental Specifications

Description Specification

Operating temperature 23 to 113°F

Non-operating temperature –13 to 158°F (–25 t

Relative humidity 5% to 95% noncondensing

(-5 to 45°C)

o 70°C)

A-2

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Appendix A ATA 190 Specifications

Physical Interfaces

Ta b l e A-4 Physical Interfaces

Description Specification

Ethernet One RJ-45 connector, IEEE 802.3 100Bas

Analog phone Two RJ-11 FXS voice ports

Power 5VDC po

Power switch Power switch to turn the ATA 190 on or off

Ringing Characteristics

Ta b l e A-5 Ringing Characteristics

Description Specification

Tip/ring interfaces for each RJ-11 FXS port (

Ring voltage 40V

Ring frequency 20 Hz

Ring waveform Trapezoidal with 1.2 to 1.6 crest factor

Ring load 1400 ohm + 40μF

Ringer equivalence number (REN) Up to 5 REN per RJ-11 FXS port

Loop impedance Up to 200 ohms (plus 430-ohm maximum phone DC

On-hook/off-hook characteristics

On-hook voltage (tip/ring) –48V

Off-hook current 24 mA (nominal)

RJ-11 FXS port terminating impedance option The ATA 190-I1 provides multiple impedance, such

wer connector

SLIC)

Physical Interfaces

eT standard

(typical, balanced ringing only)

RMS

sistance)

s 600 ohm for American SKU, 900 ohm for

a European SKU, 220 ohm (820 ohm || 120nF) for Australian SKU, and so on.

Software Specifications

Ta b l e A-6 Software Specifications (All Protocols)

Description Specification

Call progress tones Configurable for two sets of frequencies and single set of on/off

Dual-tone multifrequency (DTMF) DTMF tone detection and generation

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SIP Compliance Reference Information

Table A-6 Software Specifications (All Protocols) (continued)

Description Specification

Fax Fax pass-through and T.38 fax relay mode.

Line-echo cancellation

Out-of-band DTMF RFC 2833 AVT tones for SIP

Appendix A ATA 190 Specifications

Enhanced fax pass-through is supported on the Cisco ATA . Success of network conditions, and fax modem/fax machine tolerance to those conditions. The network must have reasonably low network jitter, network delay, and packet-loss rate.

• Echo canceller for each port

• 8 ms echo length

• Nonlinear echo suppression (ERL > 28 dB for frequency

• Convergence time = 250 ms

• ERLE = 10 to 20 dB

• Double-talk detection

fax transmissions up to 14.4 kbps depends on

= 300 to 2400 Hz)

Note Cannot transmit RFC 2833 and in-band signalling,

simultaneously.

Configuration

Quality of Service

• DHCP (RFC 2131)

• Web configuration via built-in Web server

• Touch-tone phone keypad configuration with voice prompt

• Basic boot configuration (RFC 1350 TFTP Profiling)

• Dial plan configuration

• Cisco Discovery Protocol

• Class-of-service (CoS) bit-tagging (802.1P)

• Type-of-service (ToS) bit-tagging

Security RC4 encryption for TFTP configuration files

Voice coder-decoders (codecs)

Voice features

• G.729A, G.729AB

• G.711A-law

• G.711µ-law

• Voice activity detection (VAD)

• Comfort noise generation (CNG)

• Dynamic jitter buffer (adaptive)

Voice-over-IP (VoIP) protocols SIP (RFC 3261 bis)

SIP Compliance Reference Information

Information on how the ATA 190 complies with the IETF definition of SIP as described in RFC 2543 is found at the following URL:

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Appendix A ATA 190 Specifications

http://www.ietf.org/rfc/rfc2543.txt

SIP Compliance Reference Information

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SIP Compliance Reference Information

Appendix A ATA 190 Specifications

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CHAPTER

Voice Menu Codes

This section contains information on accessing the Interactive Voice Response (IVR) and a quick-reference list of the voice configuration menu options for the ATA 190.

Accessing the IVR and Configuring Your Phone Setting

To access the IVR and configure your phone settings, follow these steps:

Note You can change the PIN on the Cisco Unified CM User Options web page.

Procedure

Step 1 To access the IVR, go off-hook on the phone connected to Line 1 and press ****.

The IVR prompts for a password.

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Note The ATA 190 allows you to enter only numerical values for password.

Step 2 Enter the PIN by pressing the number keypad, followed by # button.

You are at the IVR main configuration menu.

Step 3 Follow the voice prompts on the IVR. See Table B-1 for information on navigating the IVR.

Step 4 To return to the main configuration menu, press #.

Step 5 To exit the IVR, end the call.

Table B-1 describes the various options in the IVR Configuration Menu

Ta b l e B-1 Navigating the IVR Configuration Menu

Action IVR Code Navigating Notes

Show IP address 110

Configure IP address 111 Availaible in static ip mode only

Show subnet mask 120

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Chapter B Voice Menu Codes

Table B-1 Navigating the IVR Configuration Menu (continued)

Action IVR Code Navigating Notes

Configure subnet mask 121 Availaible in static ip mode only

Show default gateway 130

Configure default gateway 131 Availaible in static ip mode only

Show TFTP server address 220

Configure TFTP server address 221

Show LAN mode 100 Value 0 for DHCP and 1 for static IP

Configure LAN mode 101 Value 0 for DHCP and 1 for static IP

Show VLAN 230

Configure VLAN 231 To disable VLAN—Set VLAN id to 4095

To enable VLAN—Set VLAN id from 1 to 4094

Factory Reset 73738

B-2

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APPENDIX

Recommended ATA 190 Tone Parameter Values by Country

This section provides tables of recommended tone parameters for the followings countries, listed alphabetically:

Note The extended tone format used by some countries is available with ATA 190 firmware version 9.0(3).

• Australia

• Germany

• Italy

• New Zealand

• United States

Ta b l e C-1 Australia

Parameter Recommended Values

DialTone 2,31163,30958,1477,1566,1,0,0,0

BusyTone 1,30958,0,2212,0,0,3000,3000,0

ReorderTone 1,31163,0,2086,0,0,3000,3000,0

RingbackTone 102,31163,1477,30742,1654,2,3200,1600,3200,16000,0

SITTone 1,30958,0,2212,0,0,20000,4000,0

Ta b l e C-2 Germany

Parameter Recommended Values

DialTone 1,30958,0,3125,0,1,0,0,0

BusyTone 1,30958,0,1757,0,0,3840,3840,0

ReorderTone 1,30958,0,1757,0,0,1920,1920,0

RingbackTone 1,30958,0,1971,0,0,8000,32000,0

SITTone 1,30958,0,1757,0,0,1920,1920,0

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Appendix C Recommended ATA 190 Tone Parameter Values by Country

Ta b l e C-3 Italy

Parameter Recommended Values

DialTone 101,30958,3125,0,0,2,1600,1600,4800,8000,0

BusyTone 1,30958,0,1757,0,0,4000,4000,0

ReorderTone 1,30958,0,1757,0,0,1600,1600,0

RingbackTone 1,30958,0,1971,0,0,8000,32000,0

SITTone 1,30958,0,1757,0,0,4000,4000,0

Ta b l e C-4 New Zealand

Parameter Recommended Values

DialTone 1,31163,0,3307,0,1,0,0,0

BusyTone 1,31163,0,1657,0,0,4000,4000,0

ReorderTone 1,24916,0,3483,0,0,4000,4000,0

RingbackTone 102,31163,1316,30742,1474,2,3200,1600,3200,16000,0

SITTone 100,1,31163,1657,0,0,0,0,2,6000,800,6000,3200,0,0,2,0

Ta b l e C-5 United States

Parameter Recommended Values

DialTone 2,31537,30830,1490,1859,1,0,0,0

BusyTone 2,30466,28958,1246,1583,0,4000,4000,0

ReorderTone 2,30466,28958,1246,1583,0,2000,2000,0

RingbackTone 2,30830,30466,793,862,0,8000,24000,0

SITTone 2,30466,28958,1246,1583,0,2000,2000,0

C-2

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Troubleshooting and Maintenance

This chapter provides information that can assist you in troubleshooting problems with your ATA 190 or with your IP telephony network. It also explains how to clean and maintain your phone.

This chapter includes these topics:

• Resolving Startup Problems, page D-1

• ATA 190 Resets Unexpectedly, page D-5

• Troubleshooting ATA 190 Security, page D-7

• General Troubleshooting Tips, page D-7

• Where to Go for More Troubleshooting Information, page D-9

• Cleaning the ATA 190, page D-9

Resolving Startup Problems

CHAPTER

After installing an ATA 190 into your network and adding it to Cisco Unified Communications Manager, the phone should start up as described in the Installing the ATA 190, page 3-2. If the phone does not start up properly, see the following sections for troubleshooting information:

• Symptom: The ATA 190 Does Not Go Through its Normal Startup Process, page D-1

• Symptom: The ATA 190 Does Not Register with Cisco Unified Communications Manager, page D-2

• Symptom: ATA 190 Unable to Obtain IP Address, page D-4

Symptom: The ATA 190 Does Not Go Through its Normal Startup Process

When you connect a phone in the network port, the phone should go through its normal startup process as described in the Verifying the ATA 190 Startup Process, page 3-3. If the phone does not go through the startup process, the cause may be faulty cables, bad co and so on. Or, the phone may not be functional.

To determine whether the phone is functional, follow t other potential problems:

1. Verify that the network port is functional:

–

Exchange the Ethernet cables with cables that you know are functional.

nnections, network outages, lack of power,

hese suggestions to systematically eliminate these

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Chapter D Troubleshooting and Maintenance

Resolving Startup Problems

–

Disconnect a functioning phone from another port and connect it to this network port to verify the port is active.

–

Connect the phone that will not start up to a different network port that is known to be good.

–

Connect the phone that will not start up directly to the port on the switch, eliminating the patch panel connection in the office.

2. Verify that the phone is receiving power:

–

If you are using external power, verify that the electrical outlet is functional.

–

If you are using the external power supply, switch with a unit that you know to be functional.

3. If the phone still does not start up properly, perform a factory reset of the phone.

Symptom: The ATA 190 Does Not Register with Cisco Unified Communications Manager

If the phone proceeds past the first stage of the startup process (LED buttons flashing on and off) but continues to cycle through the messages, the phone is not starting up properly. The phone cannot successfully start up unless it is connected to the Ethernet network and it has registered with a Cisco Unified Communications Manager server.

These sections can assist you in d

• Registering the Phone with Cisco Unified Communications Manager, page D-4

etermining the reason the phone is unable to start up properly:

• Checking Network Connectivity, page D-2

• Verifying TFTP Server Settings, page D-2

• Verifying DNS Settings, page D-3

• Verifying Cisco Unified Communications Manager Settings, page D-3

• Cisco Unified Communications Manager and TFTP Services Are Not Running, page D-3

• Creating a New Configuration File, page D-3

• Registering the Phone with Cisco Unified Communications Manager, page D-4

Checking Network Connectivity

If the network is down between the phone and the TFTP server or Cisco Unified Communications Manager, the phone cannot start up properly. Ensure that the network is currently running.

Verifying TFTP Server Settings

You can determine the IP address of the TFTP server used by the ATA 190 by entering http://x.x.x.x where x.x.x.x is the IP address of the ATA 190.

If you have assigned a static IP address to the phone, you must manually enter a setting for the TFTP

er 1 option. See “Accessing the IVR and Configuring Your Phone Setting” section on page B-1.

Serv

If you are using DHCP, the phone obtains the address for the TFTP server from the DHCP server. Check

he IP address configured in Option 150.

D-2

You can also enable the phone to use an alte phone was recently moved from one location to another. See “Accessing the IVR and Configuring Your

Phone Setting” section on page B-1 for instructions.

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Chapter D Troubleshooting and Maintenance

Verifying DNS Settings

If you are using DNS to refer to the TFTP server or to Cisco Unified Communications Manager, you must ensure that you have specified a DNS server. Verify this setting by entering http://x.x.x.x where x.x.x.x is the IP address of the ATA 190. You should also verify that there is a CNAME entry in the DNS server for the TFTP server and for the Cisco Unified Communications Manager system.

You must also ensure that DNS is configured to do reverse look-ups.

Verifying Cisco Unified Communications Manager Settings

Enter http://x.x.x.x where x.x.x.x is the IP address of the ATA 190 to find the active Cisco Unified Communications Manager settings.

Cisco Unified Communications Manager and TFTP Services Are Not Running

If the Cisco Unified Communications Manager or TFTP services are not running, phones may not be able to start up properly. However, in such a situation, it is likely that you are experiencing a system-wide failure, and that other phones and devices are unable to start up properly.

Resolving Startup Problems

If the Cisco Unified Communications Manager service is not running, all devices on the network that

on it to make phone calls will be affected. If the TFTP service is not running, many devices will not

rely be able to start up successfully.

To start a service, follow these steps:

Procedure

Step 1 From Cisco Unified Communications Manager Administration, choose Cisco Unified Serviceability

from the Navigation drop-down list.

Step 2 Choose Tools > Control Center - Network Services.

Step 3 Choose the primary Cisco Unified Communications Manager server from the Server drop-down list.

The window displays the service names for the server th service control panel to stop or start a service.

Step 4 If a service has stopped, click its radio button and then click the Start button.

The Service Status symbol changes from a square t

Note A service must be activated before it can be started or stopped. To activate a service, choose Tools >

Service Activation.

at you chose, the status of the services, and a

o an arrow.

Creating a New Configuration File

If you continue to have problems with a particular phone that other suggestions in this chapter do not resolve, the configuration file may be corrupted.

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Resolving Startup Problems

To create a new configuration file, follow these steps:

Procedure

Step 1 From Cisco Unified Communications Manager, choose Device > Phone > Find to locate the phone

experiencing problems.

Step 2 Choose Delete to remove the phone from the Cisco Unified Communications Manager database.

Step 3 Add the phone back to the Cisco Unified Communications Manager database. See Attaching a Phone to

the ATA 190, page 3-3 for details.

Step 4 Power cycle the phone.

Note When you remove a phone from the Cisco Unified Communications Manager database, its configuration

file is deleted from the Cisco Unified Communications Manager TFTP ser number or numbers remain in the Cisco Unified Communications Manager da “unassigned DNs” and can be used for other devices. If unassigned DNs are not used by other devices, delete them from the Cisco Unified Communications Manager database. You can use the Route Plan Rep Manager Administration Guide for more information.

Chapter D Troubleshooting and Maintenance

ver. The phone’s directory

tabase. They are called

ort to view and delete unassigned reference numbers. See the Cisco Unified Communications

Note Changing the buttons on a phone button template, or assigning a different phone button template to a

phone, may result in directory numbers that are no longer accessible from the phone. The directory numbers are still assigned to the phone in the Cisco Unified Communications Manager database, but

here is no button on the phone with which calls can be answered. These directory numbers should be

t removed from the phone and deleted if necessary.

Registering the Phone with Cisco Unified Communications Manager

A phone can register with a Cisco Unified Communications Manager server only if the phone has been added to the server or if auto-registration is enabled. Review the information and procedures in the

Attaching a Phone to the ATA 190, page 3-3 to ensure that the phone has been added to the

Cisco Unified Communications Manager database.

To verify that the phone is in the Cisco Unified Communications Manager database, choose De Phone > Find from Cisco Unified Communications Manager Administration to search for the phone

ased on its MAC Address. For information about determining a MAC address, see Determining the

MAC Address of an ATA 190, page 2-7.

If the phone is already in the Cisco Unified Communications Manager database, its configuration file

ay be damaged. See Adding Users to Cisco Unified Communications Manager, page 4-6 for assistance.

Symptom: ATA 190 Unable to Obtain IP Address

If a phone is unable to obtain an IP address when it starts up, the phone may be not be on the same network or VLAN as the DHCP server, or the switch port to which the phone is connected may be disabled.

Make sure that the network or VLAN to which the and make sure that the switch port is enabled.

phone is connected has access to the DHCP server,

vice >

D-4

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Chapter D Troubleshooting and Maintenance

ATA 190 Resets Unexpectedly

If users report that their phones are resetting during calls or while idle on their desk, you should investigate the cause. If the network connection and Cisco Unified Communications Manager

tion are stable, a Cisco Unified IP Phone should not reset on its own.

connec

Typically, a phone resets if it has problems connecting to the Ethernet network or to Cisco Unified Communications Manager. These sections can help you identify the cause of a phone

ng in your network:

resetti

• Verifying Physical Connection, page D-5

• Identifying Intermittent Network Outages, page D-5

• Verifying DHCP Settings, page D-5

• Checking Static IP Address Settings, page D-6

• Verifying Voice VLAN Configuration, page D-6

• Eliminating DNS or Other Connectivity Errors, page D-6

• Troubleshooting ATA 190 Security, page D-7

ATA 190 Resets Unexpectedly

Verifying Physical Connection

Verify that the Ethernet connection to which the ATA 190 is connected is up. For example, check whether the particular port or switch to which the phone is connected is down and that the switch is not rebooting. Also make sure that there are no cable breaks.

Identifying Intermittent Network Outages

Intermittent network outages affect data and voice traffic differently. Your network might have been experiencing intermittent outages without detection. If so, data traffic can resend lost packets and verify that packets are received and transmitted. However, voice traffic cannot recapture lost packets. Rather than retransmitting a lost network connection, the phone resets and attempts to reconnect its network connection.

If you are experiencing problems with the voice network, you should investigate whether an existing

blem is simply being exposed.

pro

Verifying DHCP Settings

Follow this process to help determine if the phone has been properly configured to use DHCP:

Procedure

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Step 1 Verify that you have properly configured the phone to use DHCP. See Configuring Startup Network

Settings, page 3-3 for more information.

Step 2 Verify that the DHCP server has been set up properly.

Step 3 Verify the DHCP lease duration. Cisco recommends that you set it to 8 days.

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ATA 190 Resets Unexpectedly

The ATA 190 sends messages with request type 151 to renew the DHCP address leases. If the DHCP server expects messages with request type 150, the lease will be denied, forcing the ATA 190 to restart and request a new IP address from the DHCP server.

Checking Static IP Address Settings

If the phone has been assigned a static IP address, verify that you have entered the correct settings. See

Understanding Phone Configuration Files, page 2-3 for more information.

Verifying Voice VLAN Configuration

If the ATA 190 appears to reset during heavy network usage (for example, following extensive web surfing on a computer connected to same switch as phone), it is likely that you do not have a voice VLAN configured.

Isolating the phones on a separate auxiliary V

Chapter D Troubleshooting and Maintenance

LAN increases the quality of the voice traffic.

Eliminating DNS or Other Connectivity Errors

If the phone continues to reset, follow these steps to eliminate DNS or other connectivity errors:

Procedure

Step 1 Use the IVR to reset phone settings to their default values. See Accessing the IVR and Configuring Your

Phone Setting, page B-1 for details.

Step 2 Modify DHCP and IP settings:

a. Disable DHCP.

b. Assign static IP values to the phone. See Understanding Phone Configuration Files, page 2-3 for

instructions. Use the same default router set

c. Assign TFTP server. Use the same TFTP server used for other functioning ATA 190.

Step 3 On the Cisco Unified Communications Manager server, verify that the local host files have the correct

Cisco Unified Communications Manager server name mapped to the correct IP address.

Step 4 From Cisco Unified Communications Manager, choose System > Server and verify that the server is

referred to by its IP address and not by its DNS name.

Step 5 From Cisco Unified Communications Manager, choose Device > Phone and verify that you have

assigned the correct MAC address to this Cisco Unified IP Phone. For information about determining a

C address, see Determining the MAC Address of an ATA 190, page 2-7.

Step 6 Power cycle the phone.

ting used for other functioning ATA 190.

D-6

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Chapter D Troubleshooting and Maintenance

Troubleshooting ATA 190 Security

Table D-1 provides troubleshooting information for the security features on the ATA 190. For

information relating to the solutions for any of t information about security, see the Cisco Unified Communications Manager Security Guide.

Ta b l e D-1 ATA 190 Security Troubleshooting

Problem Possible Cause

CTL File Problems

Device authentication error. CTL file does not have a

Phone cannot authenticate CTL file. The security token that si

Phone cannot authenticate an files other than CTL file.

Phone does not register with

sco Unified Communications Manager.

Phone does not request signed configuration files. The CTL file does not contain any TFTP entries

y of the configuration

Troubleshooting ATA 190 Security

hese issues, and for additional troubleshooting

Cisco Unified Communications Manager

icate or has an incorrect certificate.

certif

gned the updated CTL

file does not exist in the CTL file on the phone.

The configuration file may not be signed by the corresponding certificate in the phone’s Trust List.

The CTL file does not contain the correct information for the Cisco Unified Communications Manager server.

certificates.

with

General Troubleshooting Tips

Table D-2 provides general troubleshooting information for the ATA 190.

Ta b l e D-2 ATA 190 Troubleshooting

Summary Explanation

Poor quality when calling mobile

ones using the G.729 protocol

Prolonged broadcast storms

se phones to reset, or be

cau unable to make or answer a call

In Cisco Unified Communications Manager, you can configure the network to use the G.729 protocol (the default is G.711). When using G.729, calls between a phone and a mobile phone will have poor voice quality. Use G.729 only when absolutely necessary.

A prolonged Layer 2 broadcast storm (lasting several minutes) on the voice VLAN may cause phones to reset, lose an active call, or be unable to initiate or answer a call. Phones may not come up until a broadcast storm ends.

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General Troubleshooting Tips

Table D-2 ATA 190 Troubleshooting (continued)

Summary Explanation

Moving a network connection from the phone to a workstation

Chapter D Troubleshooting and Maintenance

If you are powering your phone through the network connection, you must be careful if you decide to unplug the phone’s network connection and plug the cable into a desktop computer.

Caution The computer’s network card cannot receive power

through the network connection; if power comes through the connection, the network card can be destroyed. To protect a network card, wait 10 seconds or longer after unplugging the cable from the phone before plugging it into a computer. This delay gives the switch enough time to recognize that there is no longer a phone on the line and to stop providing power to the cable.

Changing the phone configuration

By default, the network configuration options are locked to prevent users from making changes that could impact their network connectivity. You must unlock the network configuration options before you can configure them.

Dual-Tone Multi-Frequency (DTMF) del

Codec mismatch between the

one and another device

When you are on a call that requires keypad input, if you press the keys too quickly, some of them might not be recognized.

The RxType and the TxType statistics show the codec that is being used for a conversation between this ATA 190 and the other device. The values of these statistics should match. If they do not, verify that the other device can handle the codec conversation or that a transcoder is in place to handle the service.

Sound sample mismatch

een the phone and another

betw device

Gaps in voice calls Check the AvgJtr and the MaxJtr statistics. A la

The RxSize and the TxSize statistics show the size of the voice packets that are being used in a conversation between this ATA 190 and the other device. The values of these statistics should match.

rge variance between these statistics might indicate a problem with jitter on the network or periodic high rates of network activity.

Loopback condition A loopback condition can occur when the following conditions are

met:

• The SW Port Configuration option in the Network

Configuration menu on the phone is set to 10 Half (10-BaseT/half duplex)

• The phone receives power from an external power supply

D-8

• The phone is powered down (the power supply is disconnected)

In this case, the switch port on the ph the following message will appear in the switch console log:

HALF_DUX_COLLISION_EXCEED_THRESHOLD

To resolve this problem, re-enable the port from the switch.

Cisco ATA 190 Analog Telephone Adaptor Administration Guide for SIP (Version 1.0)

one can become disabled and

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Chapter D Troubleshooting and Maintenance

Table D-2 ATA 190 Troubleshooting (continued)

Summary Explanation

One-way audio When at least one person in a call does not receive audio, IP

Phone call cannot be established The phone does not have a DHCP IP address

Where to Go for More Troubleshooting Information

ectivity between phones is not established. Check the

conn configurations in routers and switches to ensure that IP connectivity is properly configures.

, is unable to register to Cisco Unified Communications Manager, and shows a Configuring IP or Registering message.

Verify the following:

1. The Ethernet cable is attached.

2. The Cisco CallManager service is running on the Cisco Unified

Communications Manager server.

3. Both phones are registered to the same Cisco Unified

Communications Manager.

4. Audio server debug and capture logs are enabled for both

phones. If needed, enable Java debug.

Where to Go for More Troubleshooting Information

If you have additional questions about troubleshooting the ATA 190, several Cisco.com web sites can provide you with more tips. Choose from the sites available for your access level.

• ATA 190 Troubleshooting Resources:

http://www.cisco.com/en/US/products/hw/gatecont/

• Cisco Products and Services (Technical Support and Documentation):

http://www.cisco.com/cisco/web/support/index.html

ps514/tsd_products_support_series_home.html

Cleaning the ATA 190

To clean your ATA 190, use a soft, dry cloth to wipe the surface. Do not apply liquids or powders directly on the device. As with all non-weather-proof electronics, liquids and powders can damage the components and cause failures.

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Cleaning the ATA 190

Chapter D Troubleshooting and Maintenance

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Numerics

GLOSSARY

10BaseT

A-law

AVT tones

category-3 cable

CED tone detection

CELP

10-Mbps baseband Ethernet specification using two pairs of twisted-pair cabling (Categories 3, 4, or

5): one pair for transmitting data and the other for receiving data. 10BASET, which is part of the IEEE 802.3 specification, has a distance limit of approximately 328 feet (100 meters) per segment.

ITU-T companding standard used in the conversion between analog and digital signals in PCM systems. A-law is used primarily in European phone networks and is similar to the North American µ-law standard. See also companding and µ-law.

Out-of-bound signaling as defined in RFC 2833.

One of five grades of UTP cabling described in the EIA/TIA-586 standard. Category 3 cabling is used in 10BaseT networks and can transmit data at speeds up to 10 Mbps.

Called station identification. A three-second, 2100 Hz tone generated by a fax machine answering a call, which is used in the hand-shaking used to set the call; the response from a called fax machine to a CNG tone.

code excited linear prediction compression. Compression algorithm used in low bit-rate voice encoding. Used in ITU-T Recommendations G.728, G.729, G.723.1.

CLIP

CLIR

CNG

codec

companding

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Calling Line Identification Presentation. Shows your identity to callers with Caller ID.

Calling Line Identification Restriction. Hides your identity from callers with Caller ID.

Comfort Noise Generation

coder decoder. In Voice over IP, Voice over Frame Relay, and Voice over ATM, a DSP software algorithm used to compress/decompress speech or audio signals.

Contraction derived from the opposite processes of compression and expansion. Part of the PCM process whereby analog signal values are rounded logically to discrete scale-step values on a nonlinear scale. The decimal step number then is coded in its binary equivalent prior to transmission. The process is reversed at the receiving terminal using the same nonlinear scale. Compare with compression and expansion. See also a-law and µ-law.

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Glossary

compression

CoS

DHCP

dial peer

DNS

DSP

DTMF

The running of a data set through an algorithm that reduces the space required to store or the bandwidth required to transmit the data set. Compare with companding and expansion.

Class of service. An indication of how an upper-layer protocol requires a lower-layer protocol to treat its messages. In SNA subarea routing, CoS definitions are used by subarea nodes to determine the optimal route to establish a given session. A CoS definition comprises a virtual route number and a transmission priority field.

Dynamic Host Configuration Protocol. Provides a mechanism for allocating IP addresses dynamically so that addresses can be reused when hosts no longer need them.

An addressable call endpoint. In Voice over IP (VoIP), there are two types of dial peers: POTS and Vo I P.

Domain Name System. System used on the Internet for translating names of network nodes into addresses.

digital signal processor. A DSP segments the voice signal into frames and stores them in voice packets.

dual tone multifrequency. Tones generated when a button is pressed on a phone, primarily used in the U.S. and Canada.

E.164

endpoint

expansion

firewall

FoIP

FQDN

FSK

The international public telecommunications numbering plan. A standard set by the ITU-T which addresses phone numbers.

A SIP terminal or gateway. An endpoint can call and be called. It generates and/or terminates the information stream.

The process of running a compressed data set through an algorithm that restores the data set to its original size. Compare with companding and compression.

Router or access server, or several routers or access servers, designated as a buffer between any connected public networks and a private network. A firewall router uses access lists and other methods to ensure the security of the private network.

Fax over IP

Fully Qualified Domain (FQDN) format “mydomain.com” or “company.mydomain.com.”

Frequency shift key

GL-2

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Glossary

FXO

FXS

G.711

G.723.1

G.729A

Foreign Exchange Office. An FXO interface connects to the public switched phone network (PSTN) central office and is the interface offered on a standard phone. Cisco FXO interface is an RJ-11 connector that allows an analog connection at the PSTN central office or to a station interface on a PBX.

Foreign Exchange Station. An FXS interface connects directly to a standard phone and supplies ring, voltage, and dial tone. Cisco's FXS interface is an RJ-11 connector that allows connections to basic phone service equipment, keysets, and PBXs.

Describes the 64-kbps PCM voice coding technique. In G.711, encoded voice is already in the correct format for digital voice delivery in the PSTN or through PBXs. Described in the ITU-T standard in its G-series recommendations.

Describes a compression technique that can be used for compressing speech or audio signal components at a very low bit rate as part of the H.324 family of standards. This Codec has two bit rates associated with it: 5.3 and 6.3 kbps. The higher bit rate is based on ML-MLQ technology and provides a somewhat higher quality of sound. The lower bit rate is based on CELP and provides system designers with additional flexibility. Described in the ITU-T standard in its G-series recommendations.

Describes CELP compression where voice is coded into 8-kbps streams. There are two variations of this standard (G.729 and G.729 Annex A) that differ mainly in computational complexity; both provide speech quality similar to 32-kbps ADPCM. Described in the ITU-T standard in its G-series recommendations.

gateway

H.323

ICMP

A gateway allows SIP or H.323 terminals to communicate with terminals configured to other protocols by converting protocols. A gateway is the point where a circuit-switched call is encoded and repackaged into IP packets.

H.323 allows dissimilar communication devices to communicate with each other by using a standard communication protocol. H.323 defines a common set of CODECs, call setup and negotiating procedures, and basic data transport methods.

Internet Control Message Protocol

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Glossary

IVR

LDAP

LEC

Location Server

MWI

-law

Internet Protocol. Network layer protocol in the TCP/IP stack offering a connectionless internetwork service. IP provides features for addressing, type-of-service specification, fragmentation and reassembly, and security. Defined in RFC 791.

Interactive voice response. Term used to describe systems that provide information in the form of recorded messages over phone lines in response to user input in the form of spoken words or, more commonly, DTMF signaling.

Lightweight Directory Access Protocol

local exchange carrier

A SIP redirect or proxy server uses a location server to get information about a caller’s location. Location services are offered by location servers.

message waiting indication

North American companding standard used in conversion between analog and digital signals in PCM systems. Similar to the European a-law. See also a-law and companding.

NAT

NSE packets

NAT Server

NTP

POTS

Network Address Translation. Mechanism for reducing the need for globally unique IP addresses. NAT allows an organization with addresses that are not globally unique to connect to the Internet by translating those addresses into globally routable address spaces. Also known as Network Address Translator.

Real-Time Transport Protocol (RTP) digit events are encoded using the Named Signaling Event (NSE) format specified in RFC 2833, Section 3.0.

Network Address Translation. an Internet standard that enables a local-area network (LAN) to use one set of IP addresses for internal traffic and a second set of addresses for external traffic.

Network Time Protocol. Protocol built on top of TCP that assures accurate local time-keeping with reference to radio and atomic clocks located on the Internet. This protocol is capable of synchronizing distributed clocks within milliseconds over long time periods.

Plain old phone service. Basic phone service supplying standard single-line phones, phone lines, and access to the PSTN.

GL-4

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Glossary

Proxy Server

PSTN

QoS

Redirect Server

An intermediary program that acts as both a server and a client for the purpose of making requests on behalf of other clients. Requests are serviced internally or by passing them on, possibly after translation, to other servers. A proxy interprets, and, if necessary, rewrites a request message before forwarding it.

Public switched phone network

Quality of Service. The capability of a network to provide better service to selected network traffic over various technologies, including Frame Relay, Asynchronous Transfer Mode (ATM), Ethernet and

802.1 networks, SONET, and IP-routed networks that may use any or all of these underlying technologies. The primary goal of QoS is to provide priority including dedicated bandwidth, controlled jitter and latency (required by some real-time and interactive traffic), and improved loss characteristics.

A redirect server is a server that accepts a SIP request, maps the address into zero or more new addresses, and returns these addresses to the client. It does not initiate its own SIP request nor accept calls.

Registrar Server

router

RTP

SDP

SIP

A registrar server is a server that accepts Register requests. A registrar is typically co-located with a proxy or redirect server and may offer location services.

Network layer device that uses one or more metrics to determine the optimal path along which network traffic should be forwarded. Routers forward packets from one network to another based on network layer information. Occasionally called a gateway (although this definition of gateway is becoming increasingly outdated). Compare with gateway.

Real-Time Transport Protocol. One of the IPv6 protocols. RTP is designed to provide end-to-end network transport functions for applications transmitting real-time data, such as audio, video, or simulation data, over multicast or unicast network services. RTP provides services such as payload type identification, sequence numbering, timestamping, and delivery monitoring to real-time applications.

Session Definition Protocol. An IETF protocol for the definition of Multimedia Services. SDP messages can be part of SGCP and MGCP messages.

Session Initiation Protocol. Protocol developed by the IETF MMUSIC Working Group as an alternative to H.323. SIP features are compliant with IETF RFC 2543, published in March 1999. SIP equips platforms to signal the setup of voice and multimedia calls over IP networks.

SIP endpoint

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A terminal or gateway that acts as a source or sink of Session Initiation Protocol (SIP) voice data. An endpoint can call or be called, and it generates or terminates the information stream.

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Glossary

SLIC

SOHO

T.38

TCP

TFTP

TN power systems

TOS

Subscriber Line Interface Circuit. An integrated circuit (IC) providing central office-like phone interface functionality.

Small office, home office. Networking solutions and access technologies for offices that are not directly connected to large corporate networks.

T.38 is an ITU recommendation for allowing transmission of fax over IP networks in real time.

Transmission Control Protocol. Connection-oriented transport layer protocol that provides reliable full-duplex data transmission. TCP is part of the TCP/IP protocol stack.

Trivial File Transfer Protocol. Simplified version of FTP that allows files to be transferred from one computer to another over a network, usually without the use of client authentication (for example, username and password).

A TN power system is a power distribution system with one point connected directly to earth (ground). The exposed conductive parts of the installation are connected to that point by protective earth conductors.

Type of service. See CoS.

UAC

UAS

UDP

user agent

VAD

User agent client. A client application that initiates the SIP request.

User agent server (or user agent). A server application that contacts the user when a SIP request is received, and then returns a response on behalf of the user. The response accepts, rejects, or redirects the request.

User Datagram Protocol. Connectionless transport layer protocol in the TCP/IP protocol stack. UDP is a simple protocol that exchanges datagrams without acknowledgments or guaranteed delivery, requiring that error processing and retransmission be handled by other protocols. UDP is defined in RFC 768.

See UAS.

Voice activity detection. When enabled on a voice port or a dial peer, silence is not transmitted over the network, only audible speech. When VAD is enabled, the sound quality is slightly degraded but the connection monopolizes much less bandwidth.

GL-6

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Glossary

voice packet gateway

VoIP

XML

Gateway platforms that enable Internet telephony service providers to offer residential and business-class services for Internet telephony.

Voice over IP. The capability to carry normal telephony-style voice over an IP-based Internet with POTS-like functionality, reliability, and voice quality. VoIP enables a router to carry voice traffic (for example, phone calls and faxes) over an IP network. In VoIP, the DSP segments the voice signal into frames, which then are coupled in groups of two and stored in voice packets. VoIP is a blanket term, which generally refers to Cisco’s standard-based (for example H.323) approach to IP voice traffic.

eXtensible Markup Language. Designed to enable the use of SGML on the World-Wide Web. XML allow you to define your own customized markup language.

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Glossary

GL-8

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INDEX

Symbols

.cnf.xml configuration file 2-3

AC adapter, connecting 3-2

adding

ATA 187 manually

2-6

ATA 187 using auto-registration 2-6

Cisco Unified IP Phones using BAT 2-7

users to Cisco Unified Communications Manager 4-6

Advance Adhoc Conference service parameter 4-2

ATA 187

adding manually to Cisco Unified Communications

2-6

nager

adding to Cisco Unified Communications Manager

2-5

cleaning D-9

registering 2-5

registering with Cisco Unified Communications Manager

2-6

troubleshooting and maintenance D-1

Audible message waiting indicator 4-1

auto-registration

2-6

using

BAT (Bulk Administration Tool) 2-7

caller ID 4-2, 6-2

call forward

all calls

4-2

call forwarding

in Sweden

6-6

in United States 6-6

types 6-6

call pickup 4-2

call waiting 4-2

in Sweden 6-6

in United States 6-6

call-waiting caller ID 6-2

cbarge 4-1

Cisco Unified Communications Manager

adding phone to database of

2-5

interactions with 2-2

verifying settings D-3

Cisco Unified Communications Manager Administration

adding telephony features using

4-1

Cisco Unified IP Phone

power sources

2-2

registering with Cisco Unified Communications Manager

2-7

cleaning

cleaning the ATA 187

D-9

cleaning the ATA 187 D-9

conference 4-2

conference call

in Sweden

6-5

in United States 6-5

configuration file

.cnf.xml

2-3

creating D-3

overview 2-3

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Index

secure 2-3

configuring

user features

connecting

to AC adapter

to the network 3-2

CTL file

requesting

4-6

3-2

2-4

DHCP

troubleshooting

DHCP IP address D-9

directory numbers, assigning manually 2-6

direct transfer 4-3

DNS server

troubleshooting

verifying settings D-3

D-5

D-6

hold 4-3

installing

preparing

2-5

Lightweight Directory Access Protocol (LDAP) 1-3

meet-me conference 4-3

message waiting 4-3

methods supported 1-7

music-on-hold 4-3

electrical specifications A-2

environmental specifications A-2

fax pass-through mode 5-1

fax services 5-1

forced authorization codes 4-3

FXS ports 1-4

group call pickup 4-3

network connectivity, checking D-2

network outages, identifying D-5

Network port 1-4

network port

connecting to

3-2

physical connection, verifying D-5

physical interfaces A-3

physical specifications A-1

power

providing to the ATA 187

power source

description

external power 2-2

2-2

IN-2

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Index

privacy 4-3

proxy server 1-4

redial 4-4

redirect server 1-4

registrar server 1-4

resetting

continuously

D-4, D-5

ringing characteristics A-3

security

secure configuration file

services

1-6

sip

supplemental 1-8

shared line 4-4

SIP 1-2

clients 1-3

servers 1-3

software specifications (all protocols) A-3

speed dialing 4-1, 4-4

standard (ad hoc) conference 4-2

startup problems D-1

startup process

contacting Cisco Unified Communications

2-5

nager

loading stored ATA 187 image 2-4

obtaining IP address 2-4

obtaining power 2-4

requesting configuration file 2-5

requesting CTL file 2-4

understanding 2-4

supplementary services

common

6-1

2-3

T.38 fax relay mode 5-1

telephony features

Audible message waiting indicator

caller ID 4-2

call forward 4-2

call pickup 4-2

call waiting 4-2

cbarge 4-1

conference 4-2

direct transfer 4-3

forced authorization codes 4-3

group call pickup 4-3

hold 4-3

meet-me conference 4-3

music-on-hold 4-3

privacy 4-3

redial 4-4

shared line 4-4

speed dialing 4-4

time zone update 4-4

voice messaging system 4-4

TFTP

troubleshooting

time zone update 4-4

TLS 2-3

troubleshooting

ATA 187

Cisco Unified Communications Manager settings

D-3

DHCP D-5

DNS D-6

DNS settings D-3

network connectivity D-2

network outages D-5

physical connection D-5

services on Cisco Unified Communications Manager

D-2

D-1

D-3

4-1

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Index

TFTP settings D-2

VLAN configuration D-6

User agent client (UAC) 1-2

User agent server (UAS) 1-3

users

adding to Cisco Unified Communications

4-6

nager

using phone templates to add phones 2-7

VLAN

verifying

voice messaging system 4-4

D-6

warnings

installation

lightning activity 3-2

main disconnecting device 3-2

product disposal 3-2

3-2

IN-4

Cisco ATA 190 Analog Telephone Adaptor Administration Guide for SIP (Version 1.0)

OL-31821-01