Cisco ATA 190 Administration Manual

Cisco ATA 190 Analog Telephone Adapter Administration Guide for SIP

Version 1.0

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Text Part Number: OL-31821-01

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Cisco ATA 190 Analog Telephone Adapter Administration Guide for SIP

© 2014 Cisco Systems, Inc. All rights reserved.

Contents

C O N T E N T S

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
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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Contents

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
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
ATA 190 Resets Unexpectedly			D-5
Verifying Physical Connection				D-5

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Contents

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

G L O S S A R Y

I N D E X

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Preface

Overview

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.

Audience

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	Provides descriptions of hardware and
Adapter Overview”	software features of the ATA 190 along with a
	brief overview of the Session Initiation
	Protocol (SIP).
Chapter 2, “Preparing to Install the ATA 190	Provides information on the interactions
on Your Network”	between the ATA 190, Cisco Unified
	Communications Manager and other devices.
	It also describes options for powering the ATA
	190.
Chapter 3, “Installing the ATA 190”	Provides information on how to connect the
	ATA 190 hardware and load the QED and
	firmware files.
Chapter 4, “Configuring the ATA 190 for SIP	Provides information on how to configure the
	ATA 190 to operate with Session Initiation
	Protocol (SIP).

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Preface

		Chapter 5, “Configuring and Debugging Fax	Provides instructions for configuring both
		Services”	ports of the ATA 190 to support fax
			transmission.
		Chapter D, “Troubleshooting and	Provides basic testing and troubleshooting
		Maintenance”	procedures for the ATA 190.
		Chapter 6, “Using SIP Supplementary	Provides end-user information about pre-call
		Services”	and mid-call services.
		Chapter B, “Voice Menu Codes”	Provides a quick-reference list of the voice
			configuration menu options for the ATA 190.
		Appendix A, “ATA 190 Specifications”	Provides physical specifications for the ATA
			190.
		Appendix B, “SIP Call Flows”	Provides ATA 190 call flows for SIP scenarios.
		Appendix C, “Recommended ATA 190 Tone	Provides tone parameters for various
		Parameter Values by Country”	countries.
		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)

•Cisco ATA SIP Compliance Reference Information http://www-vnt.cisco.com/SPUniv/SIP/documents/CiscoATASIPComplianceRef.pdf

•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

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. export regulations may be found at

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.
		italic font	Arguments for which you supply values are in italics.
[ ]			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
			string will include the quotation marks.
		screen font	Terminal sessions and information the system displays are in screen font.
		boldface	Information you must enter is in boldface screen font.
		screen font
		italic screen	Arguments for which you supply values are in italic screen font.
		font
			Cisco ATA 190 Analog Telephone Adapter Administration Guide for SIP 1.0
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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 passwords are in angle brackets.

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.

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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C H A P T E R 1

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-Ethernet adapters that allow regular analog 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

Figure 1-1 Cisco Analog Telephone Adapter

390904

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Chapter 1 Cisco ATA 190 Analog Telephone Adapter Overview

Session Initiation Protocol Overview

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

SIP Capabilities

SIP provides these capabilities:

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

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

•Determines the media capabilities of the target endpoint. Using the Session Description Protocol (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 originating and target endpoint. If the call can be completed, SIP 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.

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

Session Initiation Protocol Overview

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

From an architectural standpoint, the 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 proxy and redirect servers

SIP

SIP		SIP
SIP user agents
		SIP gateway
	RTP	PSTN

Legacy PBX

72342

SIP Clients

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

SIP servers include:

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Chapter 1 Cisco ATA 190 Analog Telephone Adapter Overview

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.

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

Software Features

•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.

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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Chapter 1 Cisco ATA 190 Analog Telephone Adapter Overview

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)

•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

•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

Software Features

•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.90

•V.92

•V.44

•K56Flex

•ITU-T V.34 Annex 12

•ITU-T V.34

•V.32bis

•V.32

•V.21

•V.22

•V.23

Fax Services

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.

Note Success of fax transmission depends on network conditions and fax modem response to these conditions. 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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Chapter 1 Cisco ATA 190 Analog Telephone Adapter Overview

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

•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.

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Chapter 1 Cisco ATA 190 Analog Telephone Adapter Overview

Installation and Configuration Overview

Table 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		Installing the ATA 190, page 3-2
	ATA 190 yet.
5.	Power up the ATA 190.

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Chapter 1 Cisco ATA 190 Analog Telephone Adapter Overview

Installation and Configuration Overview

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C H A P T E R 2

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 components, including Cisco Unified Communications Manager, DNS and DHCP servers, TFTP servers, media resources, and so on.

This chapter focuses on the interactions between the ATA 190, Cisco Unified Communications Manager, 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 the Voice over IP (VoIP) network. It includes these topics:

•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

•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 Manager system before sending and receiving calls.

This section includes information on Understanding How the ATA 190 Interacts with Cisco

Unified Communications Manager, page 2-2.

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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 connected to the ATA 190, integrating traditional PBX functionality with the corporate IP network. Cisco Unified Communications Manager manages the components of the IP telephony system—the 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 described in this chapter, see Cisco Unified Communications Manager Administration Guide, Cisco Unified Communications Manager System Guide, and Cisco Unified Communications Manager Security Guide.

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

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

Unified Communications Manager that requires the phone to be reset, a change is automatically made 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 image load the phone should be running. If this 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 configuration file is set to Authenticated and the CTL file on the phone has a valid certificate for Cisco Unified Communications Manager, the phone establishes a TLS connection to Cisco Unified Communications Manager. Otherwise, the phone establishes a 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.

If you configure security-related settings in Cisco Unified 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 Cisco Unified Communications Manager Security Guide.

A phone accesses a default configuration 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 Manager database, the phone does not attempt to register with Cisco Unified Communications Manager.

If the phone has registered before, the phone accesses the 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

–Maximum length of each dial pattern is 30

The filenames are derived from the MAC Address and Description fields in the Phone Configuration window of Cisco Unified Communications Manager Administration. The MAC address uniquely identifies the phone. For more information see the Cisco Unified Communications Manager Administration Guide.

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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 configuration, not all of these process steps may occur on your ATA 190.

Table 2-1	ATA 190 Startup Process
Task	Purpose	Related Topics
1.	Obtaining Power.	See Providing Power to the ATA 190, page 2-2.
	The ATA 190 uses external power.
2.	Loading the Stored Image.
	The ATA 190 has non-volatile flash memory in which it
	stores firmware images and user-defined preferences. At
	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.
	If the ATA 190 is using DHCP to obtain an IP address, the
	device queries the DHCP server to obtain one. If you are
	not using DHCP in your network, you must assign static IP
	addresses to each device locally.
4.	Requesting the CTL file.	See the Cisco Unified Communications Manager
	The TFTP server stores the CTL file. This file contains the	Security Guide, Configuring the Cisco CTL
	certificates necessary for establishing a secure connection	Client.
	between the device and Cisco Unified Communications
	Manager.

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