Cisco Systems Spa9000 Users Manual Voice System Administration Guide

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ADMINISTRATION
GUIDE

Cisco Small Business Pro
SPA9000 Voice System Version 6.1
SPA9000 Voice System, SPA400 Internet Telephony
Gateway with 4 FXO ports and SPA9XX IP Phones

CCDE, CCSI, CCENT, Cisco Eos, Cisco HealthPresence, the Cisco logo, Cisco Lumin, Cisco Nexus, Cisco Nurse Connect, Cisco Stackpower, Cisco StadiumVision,
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All other trademarks mentioned in this document or website are the property of their respective owners. The use of the word partner does not imply a partnership relationship between
Cisco and any other company. (0903R)

© 2009 Cisco Systems, Inc. All rights reserved.

OL-17899-02

Contents

About This Document

ix
Purpose

ix

Audience

ix

Firmware

x

Organization

xi

Document Conventions

xii

Finding Information in PDF Files

xiii

Chapter 1: Getting Started

16

Introduction to the SPA9000 Voice System

16

SPA9000 IP PBX

17

SPA400 SIP-PSTN Gateway and Voicemail Server

18

IP Phones and Accessories

18

Deployment Scenarios

18

PSTN Access and Local Voice Mail

19

ITSP Service Only

20

ITSP Service, PSTN Access and Local Voice Mail

21

ITSP Service, PSTN and ISDN Access and Local Voice Mail

22

Initial Installation, and Configuration

Chapter 2: Basic Administration of the SPA9000

23

24

Upgrading Firmware for the SPA9000

25

Connecting to the SPA9000 Administration Web Server

27

Saving or Discarding Changes SPA9000

27

Access Levels

28

Setting Passwords for User and Administrator Accounts

29

SPA9000 Voice System Administration Guide

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Contents

Configuring Basic Settings

29

Setting Up the WAN Connection for the SPA9000

30

Setting the Date and Time

30

Configuring Daylight Saving Time

31

SPA9000 Ethernet Port

33

LAN and Application Guidelines

33

Configuring Multicast Addressing and Group Paging

33

Collecting System Logs and Debug Information

36

Viewing Information about the SPA9000

39

Viewing Information about Client Stations

39

Using the Interactive Voice Response Unit

40

Using the IVR Menu

40

Entering a Password through the IVR

45

Chapter 3: Configuring Your System for ITSP Interoperability

47

About the SPA9000 Voice System and SIP

47

Network Address Translation (NAT) and Voice over IP (VoIP)

49

NAT Mapping with Session Border Controller

49

NAT Mapping with SIP-ALG Router

49

Configuring NAT Mapping with a Static IP Address

49

Configuring NAT Mapping with STUN

51

Determining the Router's NAT Mechanism

53

Firewalls and SIP

54

Configuring SIP Timer Values

55

Chapter 4: Configuring Phone Lines and Calling Routing Behavior
Configuring SPA9000 FXS Ports

SPA9000 Voice System Administration Guide

56
57

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Contents

Configuring Line Interfaces on the SPA9000

58

Configuring a Line Interface for ITSP Service

58

Configuring a Line Interface for a SPA400 (PSTN or Voice Mail)

60

Configuring Call Capacity for a Line Interface

63

Configuring Dial Plans

66

About Dial Plans

66

Editing Dial Plans

74

Managing the Line Selection for Outbound Calls

78

Line Availability

78

Configuring a Call Routing Rule

79

Entering a Call Routing Rule

81

Managing Caller ID Settings for Outgoing Calls

82

Call Forwarding Support on SPA9000

82

Call Transfer Support on SPA9000

84

Call Forward Bridge Mode

84

Call Transfer Bridge Mode

84

Managing Inbound Calls with the Contact List

85

Routing an Inbound Call to the Auto Attendant

85

Routing an Inbound Call to a Receptionist or Client Stations

85

Example Contact List Rules

86

Supporting Multiple DID Numbers Per Line Interface

87

Supporting Direct Inward Dialing to Phone Extensions

88

Entering a Contact List Rule

91

Managing Inbound Calls with Hunt Groups

92

Syntax for Hunt Rules

93

Examples for Hunt Rules

94

Creating a Hunt Rule

96

Managing Inbound Calls with Shared Line Appearances

SPA9000 Voice System Administration Guide

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Contents

About Shared Line Appearances

Chapter 5: Administering the SPA400 and Voice Mail Service

98

102

Connecting to the SPA400 Administration Web Server

103

Configuring the SPA400 Network Connection

104

Saving or Discarding Changes on the SPA400

105

Managing Access to the SPA400 Configuration Utility

106

Upgrading the Firmware for the SPA400

107

Configuring a SPA400 to Interoperate with the SPA9000

108

Configuring a SPA400 for PSTN Access

111

Configuring a SPA400 for Voice Mail Service

112

Voice Mail Capacity

112

Configuring Local Voice Mail Service on a SPA400

113

Setting Up Voice Mail on Each Station

116

Enabling Remote Voice Mail Access (Optional)

119

Managing the Voice Mail Messages on the USB Key

121

Enabling Debugging on the SPA400

122

Chapter 6: Configuring Music on Hold
Using the Internal Music Source for Music On Hold

125
125

Using the Internal Music Source

125

Changing the Music File for the Internal Music Source

126

Configuring a Streaming Audio Server

127

About the Streaming Audio Server

127

Configuring the Streaming Audio Server

129

Using the IVR with an SAS Line

130

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Contents

Chapter 7: Configuring the Auto Attendant

131

How the Auto Attendant Works

131

Working with the Auto Attendant Greetings

133

Using Pre-Recorded Prompts

133

Recording an Auto Attendant Prompt

134

Downloading Prompts

137

Writing an Auto Attendant Script

138

An Introduction to XML Scripting Grammar in AA Script Examples 138
Elements of XML Scripting Grammar

142

Auto Attendant XML Instructions Set

145

Entering an Auto Attendant Script

147

Configuring the DayTime, NightTime and Weekend/Holiday Auto
Attendants
148
Configuring Dial Plans for the Auto Attendant

Chapter 8: Localization

150

151
Localizing the SPA9000 Auto Attendant Prompts

151

Local Time Configuration

154

Configuring the SPA9000 and SPA9xx Call Progress Tones

154

Localizing the SPA400 Voice Mail Prompts

160

Localizing the SPA400 Call Disconnect Tones

161

Localizing the SPA400 Caller ID Method

163

Appendix A: Advanced Topics in SPA9000 Administration
Technology Background

165
165

Session Initiation Protocol

166

SPA9000 Media Proxy

167

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Using the SPA9000 with a Firewall or Router

168

SPA400 SIP-PSTN Gateway

169

SPA9000 Architecture

170

SIP-NAT Interoperation

172

Advanced Call Control and Routing

173

Configuring Vertical (Supplementary) Service Codes

173

Managing the Outbound Call Routing Groups

175

Configuring Outbound Call Codec Selection Codes

177

Advanced Topics for SPA400 Voice Mail Service

178

How Voicemail Works

179

Checking Voicemail from an External Number

180

Depositing Voicemail

180

Subscribing to Voicemail Notification

182

Remote Provisioning Features

183

Using Configuration Profiles

183

Client Auto-Configuration

184

Manual Client Configuration

185

Client Registration

188

Using the Upgrade URL

189

Using the Resync URL

190

Using the Reboot URL

190

Appendix B: SPA9000 Field Reference
Router Tab

191
191

Status page

191

Wan Setup page

193

Lan Setup page and Application page

196

Voice tab
Info page

SPA9000 Voice System Administration Guide

197
197

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Contents

System page

200

SIP Page

202

Provisioning page

224

Regional page

224

FXS 1/2 page

241

Line 1/2/3/4 page

251

Appendix C: SPA400 Field Reference
Setup

261
261

Basic Setup

262

SPA9000 Interface

262

Voice

264

Voicemail Server

268

Voicemail Users

269

Administration

270

Management

270

Factory Default

270

USB Setting

271

Firmware Upgrade

271

Reboot

271

Status
Gateway

Event Logs
Set Log Level

Tone

Appendix D: Where to Go From Here

272
272

273
273

274

275

Product Resources

275

Related Documentation

276

SPA9000 Voice System Administration Guide

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Contents

Appendix E: Glossary

278

Appendix F: Acronyms

281

SPA9000 Voice System Administration Guide

viii

Preface

About This Document

The SPA9000 Voice System Administration Guide is intended to help VARs and
Service Providers to manage and configure the SPA9000 Voice System. This
preface provides helpful information about this guide and other resources that are
available to you. Before you begin to use this guide, refer to the following topics:

•

“Purpose,” on page ix

•

“Audience,” on page ix

•

“Firmware,” on page x

•

“Organization,” on page xi

•

“Document Conventions,” on page xii

•

“Finding Information in PDF Files,” on page xiii

Purpose
This document provides information that an administrator needs to configure the
SPA9000 Voice System, which typically consists of a SPA9000 IP PBX, one or
more SPA900 Series IP phones, and the optional SPA400 PSTN gateway and
voice mail server. This guide focuses primarily on the tasks that an administrator
performs to configure a SPA9000 with the SPA9000 administration web server.

NOTE This guide does not cover initial installation and configuration, SPA900 Series

phone configuration, the Setup Wizard, or provisioning. See “Related
Documentation” in Appendix D, on page 276.

Audience
This document is written for the following audience:

•

Service providers offering services using Cisco SPA products

•

VARs and resellers who need configuration references for Cisco SPA products

SPA9000 Voice System Administration Guide

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Preface

•

System administrators or anyone who installs and administers the SPA9000
Voice System

NOTE This guide does not provide the configuration information required by specific

service providers. Please consult with the service provider for specific service
parameters.

Firmware
This guide describes the features that are available in the following firmware
releases (and higher versions). You can find all available firmware updates by
going to Cisco.com and choosing Support. Or visit the following URL and enter the
model number in the Software Search box:
http://tools.cisco.com/support/downloads

Product

Firmware Version

SPA9000

6.1.5

SPA400

1.1.2.2

SPA901

5.1.5

SPA921/SPA941

5.1.8

SPA922/942

6.1.3

SPA962

6.1.3

WIP310

5.0.8

SPA9000 Voice System Administration Guide

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Preface

Organization
The information in this guide is organized into the following chapters and
appendices:

Chapter

Description

Chapter 1, “Getting
Started.”

This chapter introduces you to the SPA9000
Voice System by describing the components and
presenting several deployment scenarios.

Chapter 2, “Basic
Administration of the
SPA9000.”

This chapter introduces you to basic
administrative tasks using the SPA9000
administration web server and the Interactive
Voice Response Unit.

Chapter 3, “Configuring
Your System for ITSP
Interoperability”

This chapter provides configuration details to help
you to ensure that your infrastructure properly
supports voice services.

Chapter 4, “Configuring
Phone Lines and Calling
Routing Behavior”

This chapter describes many features that you
can configure on the SPA9000 to ensure smooth
handling of all inbound and outbound calls, and
ease of use.

Chapter 5, “Administering
the SPA400 and Voice Mail
Service”

This chapter guides you through the process of
configuring and managing the SPA400 for PSTN
access and voice mail service.

Chapter 6, “Configuring
Music on Hold”

This chapter explains how to configure Music on
Hold using either a music file or streaming audio.

Chapter 7, “Configuring the
Auto Attendant”

This chapter describes how to configure the
SPA9000 Voice System Auto Attendant (AA) by
using the IVR and XML scripting.

Chapter 8, “Localization”

This chapter explains how to localize your
SPA9000 Voice System with the language files,
tones, and ring patterns for your region.

Appendix A, “Advanced
Topics in SPA9000
Administration”

This appendix provides more detailed technical
information for administrators who want to
understand how the SPA9000 Voice System
works.

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Preface

Chapter

Description

Appendix B, “SPA9000
Field Reference”

This appendix describes the fields on each page
of the SPA9000 administration web server.

Appendix C, “SPA400 Field
Reference”

This appendix describes the fields on each page
of the SPA400 administration web server.

Appendix D, “Where to Go
From Here”

This appendix describes additional resources
that are available to help you and your customer
obtain the full benefits of the SPA9000 Voice
System.

Appendix E, “Glossary”
Appendix F, “Acronyms”

These resources help you to understand the
terms and acronyms that are used in this guide.

Document Conventions
The following table describes the typographic conventions that are used in this
document.
Typographic
Element

Meaning

Boldface

May indicate either of the following:

Italic

•

A user interface element that you need to click, select, or
otherwise act on

•

A literal value to be entered in a field.

May indicate either of the following:

•
•
Monospaced
Font

SPA9000 Voice System Administration Guide

A variable that should be replaced with a literal value.
The name of a page, section, or field in the user interface

Indicates code samples or system output.

xii

Preface

Finding Information in PDF Files
The SPA9000 Voice System documents are published as PDF files. The PDF Find/
Search tool within Adobe® Reader® lets you find information quickly and easily
online. You can perform the following tasks:

•

Search an individual PDF file.

•

Search multiple PDF files at once (for example, all PDFs in a specific folder or
disk drive).

•

Perform advanced searches.

Finding Text in a PDF
Follow this procedure to find text in a PDF file.
STEP 1

Enter your search terms in the Find text box on the toolbar.

NOTE By default, the Find tool is available at the right end of the Acrobat toolbar. If

the Find tool does not appear, choose Edit > Find.

STEP 2

Optionally, click the arrow next to the Find text box to refine your search by
choosing special options such as Whole Words Only.

STEP 3

Press Enter.

STEP 4

Acrobat displays the first instance of the search term.

STEP 5

Press Enter again to continue to more instances of the term.

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Preface

Finding Text in Multiple PDF Files
The Search window lets you search for terms in multiple PDF files that are stored
on your PC or local network. The PDF files do not need to be open.
STEP 1

Start Acrobat Professional or Adobe Reader.

STEP 2

Choose Edit > Search, or click the arrow next to the Find box and then choose
Open Full Acrobat Search.

STEP 3

In the Search window, complete the following steps:
a. Enter the text that you want to find.
b. Choose All PDF Documents in.
From the drop-down box, choose Browse for Location. Then choose the
location on your computer or local network, and click OK.
c. If you want to specify additional search criteria, click Use Advanced Search
Options, and choose the options you want.
d. Click Search.

SPA9000 Voice System Administration Guide

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Preface

STEP 4

When the Results appear, click + to open a folder, and then click any link to open
the file where the search terms appear.

For more information about the Find and Search functions, see the Adobe Acrobat
online help.

SPA9000 Voice System Administration Guide

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1
Getting Started

This chapter introduces you to the SPA9000 Voice System by describing the
components and presenting several deployment scenarios.

NOTE This chapter is essential reading before you begin installing the equipment or

configuring the system.

•

“Introduction to the SPA9000 Voice System,” on page16

•

“Deployment Scenarios,” on page18

•

“Initial Installation, and Configuration,” on page 23

Introduction to the SPA9000 Voice System
The SPA9000 Voice System is an affordable and feature-rich IP telephone system
that is designed especially for the Small and Home Office. The SPA9000 Voice
System uses standard TCP/IP protocols and can provide global connectivity
through any Internet Telephony Service Provider (ITSP) that supports the Session
Initiation Protocol (SIP).
At minimum, the SPA9000 Voice System includes a SPA9000 IP PBX and one or
more SPA900 series IP phones. These devices are connected through a switch to
a local area network. With an Internet connection, the SPA9000 Voice System can
subscribe to ITSP services to take advantage of low calling rates. With the
SPA400, the SPA9000 Voice System can connect to the Public Switched
Telephone Network (PSTN) to support analog phone lines. See Figure 1 “SPA9000
Voice System with the SPA9000 and SPA400” on page17 to learn more about a
typical deployment.

SPA9000 Voice System Administration Guide

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Getting Started
Introduction to the SPA9000 Voice System

Figure 1

1

SPA9000 Voice System with the SPA9000 and SPA400

SPA9000 IP PBX
The SPA9000 is an IP PBX that supports up to 16 phones. It also has a built-in
Analog Telephone Adapter (ATA) with two FXS ports for analog telephones, fax
devices, or an external music source for the music on-hold service. Devices
connected to the FXS ports are not included in the device count.
The SPA9000 has four line interfaces, which can be configured in any combination
for ITSP service, ISDN access, SPA400 PSTN access, or SPA400 voice mail
service. A different ITSP account can be configured on each line interface. If a
service provider supplies a group of sequential direct inward dial (DID) phone
numbers (such as 408-555-0100 through 555-0145) the SPA9000 can support all
of the assigned numbers on a single line interface.

SPA9000 Voice System Administration Guide

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1

Getting Started
Deployment Scenarios

The SPA9000 includes an Auto Attendant service that plays pre-recorded voice
messages to offer the caller a menu of choices and to direct the call. When the
Auto-Attendant is enabled, it parses and operates on user key presses according
to the rules that are specified in the Auto Attendant script.

SPA400 SIP-PSTN Gateway and Voicemail Server
The SPA400 provides a SIP-PSTN gateway for voice connectivity between the
PSTN and the local client stations that are connected to the SPA9000. It also
includes an integrated voice mail application that supports up to 32 voice mail
accounts with customized greetings, providing the ability to receive and playback
voice mail messages.
Each SPA400 occupies one of the four line interfaces on the SPA9000. The
SPA400 has four ports for that can be connected to PSTN or ISDN lines.

IP Phones and Accessories
The SPA9000 Voice System supports any of the Cisco SPA900 Series SIP IP
Phones, as well as the Cisco WIP310 Wireless IP Phone.

NOTE This guide explains how to configure the SPA9000 and the SPA400 to support the

calling features on the phones. For more information about the phones, see the
SPA9x2 Phone Administration Guide, the SPA9x2 Phone User Guide, and the
Cisco Wireless-G IP Phone User Guide.

Deployment Scenarios
The SPA9000 Voice System can meet the calling needs of many small businesses.
Various deployment scenarios are possible. This section includes the following
examples:

•

“PSTN Access and Local Voice Mail,” on page19

•

“ITSP Service Only,” on page 20

•

“ITSP Service, PSTN Access and Local Voice Mail,” on page 21

•

“ITSP Service, PSTN and ISDN Access and Local Voice Mail,” on page 22

SPA9000 Voice System Administration Guide

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1

Getting Started
Deployment Scenarios

PSTN Access and Local Voice Mail
In this scenario, the customer requires a robust phone system but is not using VoIP
services. The SPA9000 Voice System is deployed with a SPA9000 IP PBX, one
SPA400 for PSTN access with four FXO ports, and another SPA400 for local voice
mail service. Up to 16 IP phones can be installed. Optionally, analog phones or fax
machines (not illustrated) can be connected to the two phone ports on the
SPA9000.

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1

Getting Started
Deployment Scenarios

ITSP Service Only
In this scenario, a customer has no legacy telephone numbers and either needs no
voice mail at all or has voice mail hosted by the ITSP. The SPA9000 Voice System
is deployed with the SPA9000 IP PB and VoIP service. Up to 16 IP phones can be
installed. Optionally, analog phones or fax machines (not illustrated) can be
connected to the two phone ports on the SPA9000.

SPA9000 Voice System Administration Guide

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1

Getting Started
Deployment Scenarios

ITSP Service, PSTN Access and Local Voice Mail
In this scenario, the customer wants to use ITSP service for reduced long distance
fees but needs to support legacy local telephone numbers (for example, to receive
calls to a legacy telephone number or to place outbound calls in the local area).
This customer also prefers local voice mail service. The SPA9000 Voice System is
deployed with the SPA9000 IP PBX, VoIP service, one SPA400 unit for voice mail
service, and another SPA400 unit for PSTN access with four FXO ports. Up to 16 IP
phones can be installed. Optionally, analog phones or fax machines (not illustrated)
can be connected to the two phone ports on the SPA9000.

SPA9000 Voice System Administration Guide

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1

Getting Started
Deployment Scenarios

ITSP Service, PSTN and ISDN Access and Local Voice Mail
In this scenario, the customer takes full advantage of the SPA9000 Voice System
solution. This customer has the SPA9000 IP PBX, VoIP service, one SPA400 unit for
voice mail service, and another SPA400 for PSTN access with four FXO ports. In
addition, this installation includes an ISDN Gateway for ISDN BRI access with four
BRI ports. Up to 16 IP phones can be installed. Optionally, analog phones or fax
machines (not illustrated) can be connected to the two phone ports on the
SPA9000.

SPA9000 Voice System Administration Guide

22

Getting Started
Initial Installation, and Configuration

1

Initial Installation, and Configuration
Cisco strongly recommends that you use the SPA9000 Voice System Installation
and Configuration Guide to design your system, to prepare the site, to connect
and configure your equipment, and to set up the essential calling features. By
following the instructions in the installation guide, you can get your system running
in less time and with the settings that help to ensure strong performance.
After you complete the procedures in the installation guide, the users can make
and receive calls. When the optional SPA400 is installed, the users also can record
and retrieve voice mail messages. The SPA9000 has a fully functional Auto
Attendant to greet callers, and a default dial plan that is suitable for most dialing
scenarios. You can use this administration guide to refine the settings, to configure
advanced features, and to manage the system.

NOTE Because the SPA9000 Voice System Installation and Configuration Guide

provides all of the procedures that you need for initial installation and configuration,
those instructions are not duplicated in this administration guide.

SPA9000 Voice System Administration Guide

23

2
Basic Administration of the SPA9000

This chapter introduces you to basic administrative tasks using the SPA9000
administration web server and the Interactive Voice Response Unit.

NOTE This administration guide does not cover the initial installation and configuration of

the system. For information about connecting the equipment to start using your
system, see the SPA9000 Voice System Installation and Configuration Guide.
See the following topics:

•

“Upgrading Firmware for the SPA9000,” on page 25

•

“Connecting to the SPA9000 Administration Web Server,” on page 27

•

“Saving or Discarding Changes SPA9000,” on page 27

•

“Access Levels,” on page 28

•

“Setting Passwords for User and Administrator Accounts,” on page 29

•

“Configuring Basic Settings,” on page 29

•

“Viewing Information about the SPA9000,” on page 39

•

“Viewing Information about Client Stations,” on page 39

•

“Configuring Multicast Addressing and Group Paging,” on page 33

•

“Using the Interactive Voice Response Unit,” on page 40

SPA9000 Voice System Administration Guide

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Basic Administration of the SPA9000
Upgrading Firmware for the SPA9000

2

Upgrading Firmware for the SPA9000
As needed, you can download new firmware and then install it on the SPA9000.
STEP 1

Download the latest SPA9000 firmware from the following URL:
http://tools.cisco.com/support/downloads/go/Redirect.x?mdfid=282414116

STEP 2

Extract the Zip file, and then run the executable file to upgrade the firmware. When
the Firmware Upgrade Warning window appears, click Continue.

STEP 3

In the next window that appears, enter the IP address of the SPA9000, and then
click OK.

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Basic Administration of the SPA9000
Upgrading Firmware for the SPA9000

2

STEP 4

In the Confirm Upgrade window, verify that the correct device information and
product number appear. Then click Upgrade.

STEP 5

When the confirmation message appears, click OK.

STEP 6

To verify the upgrade, complete the following steps:
a. Connect to the SPA9000 administration web server, and choose Admin access
with Advanced settings. (See “Connecting to the SPA9000 Administration Web
Server,” on page 27).
b. Review the Router > Status page. The Software Version field should show the
firmware version that you installed.

NOTE You may need to refresh your browser to display the updated page

reflecting the new version number.

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Basic Administration of the SPA9000
Connecting to the SPA9000 Administration Web Server

2

Connecting to the SPA9000 Administration Web Server
To connect to the administration web server, perform the following steps.

NOTE You should have previously connected and configured the SPA9000 as described

in the SPA9000 Voice System Installation Guide.

STEP 1

Start Internet Explorer on a computer that is on the same network as the SPA9000.

STEP 2

Enter the IP address of the SPA9000.

NOTE To find the IP address of the SPA9000, connect an analog telephone to the
Phone 1 or Phone 2 port on the SPA9000. Then lift the receiver of the phone

and press **** on the keypad to access the IVR menu. Press 110# to hear the
IP address.
STEP 3

To view administrative features, click Admin Login and then click Advanced. By
default, no password is required. For more information, see “Setting Passwords for
User and Administrator Accounts,” on page 29.
ALTERNATIVELY: After starting Internet Explorer, enter: /
admin/advanced

Saving or Discarding Changes SPA9000
Changes can be saved or discarded at any time.

•

Changes are submitted only when you click the Submit All Changes button at
the bottom of a page. When changes are saved, the SPA9000 may reboot,
depending on the type of changes.

•

To discard unsubmitted changes, click the Undo All Changes button at the
bottom of the page.

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Basic Administration of the SPA9000
Access Levels

2

•

Unsubmitted changes are retained when you move among the pages within
the Voice module or the Router module. This feature allows you to make
changes on various pages within a module before clicking Submit All
Changes.

•

Unsubmitted changes are discarded when you switch between the Router and
Voice tabs, between the User and Administrator accounts, or between the
Basic and Advanced views.

Before you make changes, it is recommended that you save a copy of your current
working configuration:
STEP 1

In Internet Explorer, connect to the administration web server.

STEP 2

From the menu, choose File > Save As.

STEP 3

Save the configuration as Web Page Complete. You can use the saved file to
review the saved settings in all pages of the administrative GUI.

NOTE To save a Telephone Configuration, first enter the IP address for the

configuration, and then follow the above procedure.

Access Levels
You can use the SPA9000 administration web server to configure and manage
your system. Three levels of access are available:

•

User Level: The User account only has the privilege to access part of the web
profile parameters.

•

Administrator Level: The Administrator account has the privilege to modify all
the web profile parameters and can also modify the passwords of both
Administrator and User account.

•

Advanced: Administrators and Users can view advanced features by clicking
the Advanced link in the top right corner or lower left corner of the menu bar.

By default, no passwords are assigned for either the Administrator account or the
User account. If the password has been set for the Administrator account, the
browser prompts for authentication.
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Setting Passwords for User and Administrator Accounts

2

You can switch from User access to Administrator access by clicking the Admin
Login link. Likewise, you can switch from Administrator access to User access by
clicking the User Login link. If a password is set, you will be prompted to enter the
password after you click the link.

Setting Passwords for User and Administrator Accounts
The Administrator account name for the SPA9000 is admin and the User account
name is user. These account names are case sensitive and cannot be changed.

NOTE The system prompts for an Administrator account password only if a password has

been set. By default, there is no password. You should set a password to protect
your SPA9000 from unauthorized access.

STEP 1

Connect to the SPA9000 administration web server, and choose Admin access
with Advanced settings. (See “Connecting to the SPA9000 Administration Web
Server,” on page 27.)

STEP 2

Click Voice tab > System.

STEP 3

In the System Configuration section, enter the Admin Password and the User
Password, as needed. Up to 39 characters are allowed for the passwords.

STEP 4

Click Submit All Changes.

Configuring Basic Settings
This section provides information about the following tasks:

•

“Setting Up the WAN Connection for the SPA9000,” on page 30

•

“Setting the Date and Time,” on page 30

•

“Configuring Daylight Saving Time,” on page 31

•

“LAN and Application Guidelines,” on page 33

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Configuring Basic Settings

•

“Configuring Multicast Addressing and Group Paging,” on page 33

•

“Collecting System Logs and Debug Information,” on page 36

2

Setting Up the WAN Connection for the SPA9000
The SPA9000 becomes a DHCP client of any server on the network. The
recommended setting is to use a static IP address. This configuration provides
ease of installation and prevents connectivity issues that would occur if the IP
address of the SPA9000 changed.
STEP 1

Connect to the SPA9000 administration web server, and choose Admin access
with Advanced settings. (See “Connecting to the SPA9000 Administration Web
Server,” on page 27.)

STEP 2

Click Router tab > Wan Setup.

STEP 3

From the Connection Type drop-down list, choose Static IP.

STEP 4

In the Static IP Settings area, enter the Static IP of the SPA9000, as well as the
NetMask and Gateway for your network.

STEP 5

In the Optional Settings area, enter the Primary DNS for your network.

NOTE It is recommended to set an IP address that is outside the address range

assigned by the DHCP server. For example, if the DHCP server assigns IP
addresses in the range from 192.168.1.50 to 192.168.1.254, you should select
a static IP address between 192.168.1.2 and 192.168.1.49.
STEP 6

Click Submit All Changes. The SPA9000 reboots.

Setting the Date and Time
The date and time appear on the phone display and are used to activate the
daytime and nighttime Auto Attendant settings. Normally the date and time are set
by the network, which has a connection to an NTP server. If needed, you can
identify the NTP server on the Voice > Wan Setup page, Optional Settings
section.

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2

NOTE Do not use the date/time settings on the Voice > Regional page to set your system

time.

STEP 1

Connect to the SPA9000 administration web server, and choose Admin access
with Advanced settings. (See “Connecting to the SPA9000 Administration Web
Server,” on page 27).

STEP 2

Click Router tab > Wan Setup.

STEP 3

Scroll down to the Optional settings section.

STEP 4

Enter the fully qualified domain name of the NTP server that you want to use, such
as time.nist.gov.

STEP 5

Click Submit All Changes.

STEP 6

Click Voice tab > Regional.

STEP 7

Scroll down to the Miscellaneous section.

STEP 8

From the Time Zone drop-down list, choose your local time zone.

STEP 9

Click Submit All Changes.

Configuring Daylight Saving Time
You can enter a daylight saving time rule to ensure that the time is adjusted
appropriately for your region.

Syntax and Examples
SYNTAX: start = ; end=; save = 
EXAMPLE: start=3/9/7;end=11/2/7;save=1
In this example, Daylight Saving Time begins March 9, 2007, and ends Nov. 2, 2007.
One hour is added to the time of day during this period.

•

: The start date/time of daylight saving time

•

: The end date/time of daylight saving time

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2

Enter these values in the following format: //[/
HH[:mm[:ss]]]

•

•

: 1-12 (January-December)

•

: 1-31

•

: Optional. If included, this value causes the rule to take effect on
a particular day of the week before or after the specified date. Use the
values 1-7 to represent the days Monday (1) through Sunday (7). Omit this
parameter or enter 0 to cause the rule to take effect exactly on the specified
date. If  is not 0 and the  value is positive, then daylight
saving time starts or ends on  on or after the specified date. If
 is not 0 and the  value is negative, then daylight saving
time starts or ends on  on or before the specified date.

•

HH:mm:ss: Optional. The time of day when the setting takes effect, in hours
(0-23), minutes (0-59), and seconds (0-59)

: The number of hours (and optionally minutes and/or seconds)
to add to the NTP server time during daylight saving time. Enter a negative (-)
sign before  if subtraction is desired instead of addition.

Entering the Daylight Saving Time Rule
Follow this procedure to configure daylight saving time on your SPA9000 Voice
System.
STEP 1

Connect to the SPA9000 administration web server, and choose Admin access
with Advanced settings. (See “Connecting to the SPA9000 Administration Web
Server,” on page 27).

STEP 2

Click Voice tab > Regional.

STEP 3

Scroll down to the Miscellaneous section.

STEP 4

Enter the rule in the Daylight Saving Time Rule field.

STEP 5

Click Submit All Changes.

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SPA9000 Ethernet Port
The SPA9000 Ethernet port is used to connect an administrative computer.
Typically, this port is used only during initial installation and configuration. With
WAN access enabled by default, you can manage your SPA9000 from any
computer that is connected to the same subnetwork as the SPA9000. The default
IP address for this port is 192.168.0.1.

LAN and Application Guidelines
Although the SPA9000 can provide router and application services, it does not
have sufficient power to provide both phone and routing/application services in a
highly utilized environment. For this reason, Linksys recommends that the
SPA9000 not be used as a router at any time. Instead, use the SPA9000 as an
appliance by connecting its INTERNET port to a network switch and leaving the
ETHERNET port disconnected.
It is recommended that you leave the LAN and Application settings at the default
values.

Configuring Multicast Addressing and Group Paging
For initialization and system updates, the SPA9000 communicates with all the
client stations at once by using IP multicast. This method also is used in the group
paging application. For this reason, the SPA9000 and the SPA9xx IP phones must
reside on a network where multicasting is allowed. Default addresses are
provided, but you can change these addresses as needed.

NOTE Make sure that the SPA9000 and the SPA900 Series phones use the same

multicast address and port number. Also make sure that you enable spanning tree
and port fast on your LAN switch, as described in the SPA9000 Voice System
Installation and Configuration Guide.

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Configuring Basic Settings

Setting the Multicast Address
For administration purposes, the SPA9000 can send the following reboot, restart,
page, and ring messages to the group:

•

Graceful reboot: The device reboots when there are no calls in progress.

•

Immediate reboot: The device reboots immediately.

•

Graceful restart: The device restarts when there are no calls in progress.

•

Immediate restart: The device restarts immediately.

•

Group page start: One-way audio is sent from the caller to all other phones.

•

Group page end: An active page is terminated.

•

Get ringing calls: When a user chooses Group Pickup on a phone, the SPA9000
gathers information about all ringing phones and reports this information to the
requesting phone.

STEP 1

Connect to the SPA9000 administration web server, and choose Admin access
with Advanced settings. (See “Connecting to the SPA9000 Administration Web
Server,” on page 27).

STEP 2

Click Voice tab > SIP.

STEP 3

Scroll down to the PBX Parameters section.

STEP 4

Enter the correct multicast address in the Multicast Address field.
Voice tab > SIP > PBX Parameters Section

NOTE The default value is 224.168.168.168:6061.

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STEP 5

Click Submit All Changes.

STEP 6

Enter the same multicast address in the phone configurations:

2

a. Click the PBX Status link to view a list of all phones.
b. Find the phone that you want to configure, and then click the hyperlink in the IP
Address column. The Telephone Configuration page appears in a separate
browser window.
c. Click the SIP tab.
d. Scroll down to the Linksys Key System Parameters section.
e. Enter the IP address in the Multicast Address field.
f.

Click Submit All Changes.

g. Click the Back button on the Internet Explorer toolbar to return to the list of
phones.
h. Repeat these steps for each phone.

Setting the Group Page Address
In the group paging application, the originator sends RTP packets to an IP
multicast address at which all the other client stations are listening. This address is
chosen by the SPA9000 and is configured on the Voice > SIP page, PBX
Parameters section, Group Page Address field.
The originator starts the group page by choosing PageGroup from the Corporate
Directory on the phone, or by using a speed dial or personal directory entry. All
client stations are alerted at once. If the client station is on a call when a group
page starts, the call is automatically placed on hold. The speaker on each paged
station is turned on automatically unless the handset or headset is being used.
Group page is one-way only. The paged client stations can only listen to the call
from the originator.
STEP 1

Connect to the SPA9000 administration web server, and choose Admin access
with Advanced settings. (See “Connecting to the SPA9000 Administration Web
Server,” on page 27).

STEP 2

Click Voice tab > SIP.

STEP 3

Scroll down to the PBX Parameters section.

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STEP 4

Enter the correct multicast address in the Group Page Address field.
Voice tab > SIP > PBX Parameters section

NOTE The default value is 224.168.168.168:34567.

STEP 5

Click Submit All Changes.

Collecting System Logs and Debug Information
If you are working with an ITSP that needs more information to configure
interoperability, you can collect system logs and debug information for the
SPA9000. You can send these logs to the ITSP for their use.
Requirements:

•

You need a PC that is on the same subnetwork as the SPA9000, to capture the
log files. This PC needs to be running a syslog daemon. Enter the IP address of
this PC on the Voice > System page, in the Syslog Server and Debug Server
fields.

•

You can deploy a syslog server to receive syslog messages from the device,
which acts as a syslog client. The syslog client device uses the syslog protocol
to send messages, based on its configuration, to a syslog server. The syslog
messages can be accessed by reviewing the "syslog.514.log" file which
resides in the same directory as the slogsrv.exe syslog server application.

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Configuring Basic Settings

2

Partners can download the Syslog Server for SPA Devices by going to Cisco
Partner Central, Voice & Conferencing page, Technical Resources section. Use the
following URL:
/www.cisco.com/web/partners/sell/smb/products/
voice_and_conferencing.html#~vc_technical_resources

NOTE As a best practice, enable logging only when needed, and disable logging when

you finish the investigation. Logging information can impact system performance.

STEP 1

Connect to the SPA9000 administration web server, and choose Admin access
with Advanced settings. (See “Connecting to the SPA9000 Administration Web
Server,” on page 27).

STEP 2

Click Voice tab > System.

STEP 3

In the Miscellaneous Settings section, enter the following settings:

•

Syslog Server: Enter the server IP address and port to collect basic
information about system activity (no SIP messages).

•

Debug Server: Enter the server IP address and port to collect information
about SIP messages.

NOTE SIP logging is not enabled until you complete this procedure by enabling

system logging on the line interface.

•

Debug Level: Choose 3 to enable debugging.

STEP 4

Click Voice tab > Line N, where N represents the line interface number of the line
that you are investigating.

STEP 5

Scroll down to the SIP Settings section, and then choose a SIP Debug Option,
based on the level of SIP information that you want to collect.
Typically, your ITSP support personnel will tell you what type of information they
need in the logs. The drop-down list includes three categories of options: none,
1-line, and full.

•

none: Disables SIP logging

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2

•

1-line: Identifies the SIP message type but does not include the message body
Options within this category allow you to choose to exclude OPT (OPTIONS
request/response), NTFY (NOTIFY request/response), and REG (REGISTER
request/response) information to reduce the length of the logs.

•

full: Includes the SIP message body
Options within this category allow you to choose to exclude OPT (OPTIONS
request/response), NTFY (NOTIFY request/response), and REG (REGISTER
request/response) information to reduce the length of the logs.

EXAMPLES:

•

If you are troubleshooting a problem with line registration, select full to include
the OPTION, NOTIFY, and REGISTER information in the logs.

•

If you are troubleshooting a call problem, select full excl. OPT|NTFY|REG. You
do not need the OPT, Notify, and Registration information to troubleshoot a call
problem.

STEP 6

Click Submit All Changes. The information is stored on the specified server and
port, with a file name in the following format: syslog.port.log.

STEP 7

IMPORTANT: When you finish collecting the information, disable the logging:
a. Click Voice tab > Line. Change SIP Debug Option to none.
b. Click Voice tab > System. In the Miscellaneous Settings section, change
Debug Level to 0.

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Basic Administration of the SPA9000
Viewing Information about the SPA9000

Viewing Information about the SPA9000
The Router Status page provides information about software version, hardware
version, MAC address, WAN connection type, IP address, and the packets that
have been sent and received.
SPA9000 Router > Status

Viewing Information about Client Stations
The PBX Status page provides information about the client stations (IP phones),
with hyperlinks to station configuration pages.
STEP 1

Connect to the SPA9000 administration web server, and choose Admin access
with Advanced settings. (See “Connecting to the SPA9000 Administration Web
Server,” on page 27).

STEP 2

To view the status information for the client stations, click the PBX Status link in the
top right corner or lower left corner of the page. The list of client stations appears.
SPA9000 > PBX Status

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Basic Administration of the SPA9000
Using the Interactive Voice Response Unit

STEP 3

2

To view the Telephone Configuration page for any station, click the hyperlink in
the IP Address column. For information about the telephone configurations, see
the Linksys Phone Administration Guide.

Using the Interactive Voice Response Unit
In addition to the administration web server, the SPA9000 is equipped with an
Interactive Voice Response unit (IVR) that allows you to perform certain
administrative tasks by using an analog phone that is connected to the SPA9000.

•

“Using the IVR Menu,” on page 40

•

“Entering a Password through the IVR,” on page 45

Using the IVR Menu
To use the IVR menu, complete the following steps.
STEP 1

Connect an analog telephone to the Phone 1 or Phone 2 port of the SPA9000.

STEP 2

Press **** (quickly press the star key four times).

STEP 3

Wait until you hear “Linksys configuration menu.”

STEP 4

Refer to Table1 ‘IVR Options” on page 41 to identify the required option.

STEP 5

Enter the required option followed by the # (pound) key.

•

To enter a period, use the star key (*).

•

When entering a value, such as an IP address, to exit without entering any
changes, press the * (star) key twice within half a second. Otherwise, the * is
treated as a decimal point.

•

After entering a value, such as an IP address, press the # (pound) key to
indicate you have finished your selection.

•

To save a new setting, press 1. To review a new setting, press 2. To re-enter a
setting, press 3. To cancel your entry and return to the main menu, press *
(star).

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Using the Interactive Voice Response Unit

For example, to enter the IP address 191.168.1.105 by keypad, press the
following keys: 191*168*1*105. Press the # (pound) key to indicate that you
have finished entering the IP address. Then press 1 to save the IP address, or
press the * (star) key to cancel your entry and return to the main menu.

•
STEP 6

If the menu is inactive for more than one minute, the SPA9000 times out. You
need to re-enter the menu by pressing ****.

To exit the menu, hang up the telephone.
The settings that you have saved take effect after you hang up the telephone. The
SPA9000 may reboot at this time.

Table 1

IVR Options

The following table shows the codes that you enter to complete various tasks in
the IVR.

IVR Action

IVR
Menu
Choice

Parameters

Notes

Enter IVR Menu

****

None

Ignore SIT or other tones
until you hear, “Linksys
configuration menu. Please
enter option followed by
the pound key or hang-up
to exit.”

Exit IVR Menu

3948

None

Check DHCP

100

None

The IVR spells "S,T,A,T,I,C" if
the setting is for a static IP
address or "D,H,C,P" for a
DHCP IP address.

Enable/Disable
DHCP

101

Enter 0 to enable

Requires password

Check WAN IP
Address

110

SPA9000 Voice System Administration Guide

Enter 1 to disable
None

IVR announces the current
IP address of the WAN port.

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IVR Action

IVR
Menu
Choice

Parameters

Notes

Set Static IP
Address

111

Enter IP address
using numbers on
the telephone key
pad. Use the * (star)
key when entering a
decimal point.

DHCP must be disabled
first, or this value is
considered an “Invalid
Option.” Hang up the phone
after setting the IP address.
The SPA9000 reboots and
the new address takes
effect. Do not attempt to
use IVR option 110
immediately after changing
the IP address. The old IP
address is reported until
the SPA9000 reboots.
Requires password

Check Network
Mask

120

None

IVR announces the current
network mask of SPA.

Set Network
Mask

121

Enter value using
numbers on the
telephone key pad.
Use the * (star) key
when entering a
decimal point.

DHCP must be disabled
first, or this value is
considered an “Invalid
Option.”
Requires password

Check Static
Gateway IP
Address

130

None

IVR announces the current
gateway IP address of SPA.

Set Static
Gateway IP
Address

131

Enter IP address
using numbers on
the telephone key
pad. Use the * (star)
key when entering a
decimal point.

DHCP must be “Disable,”
otherwise you hear, “Invalid
Option,” if you try to set this
value.

None

IVR announces the MAC
address of SPA in hex
string format.

Check MAC
Address

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Requires password

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IVR Action

IVR
Menu
Choice

Parameters

Notes

Check Firmware
Version

150

None

IVR announces the version
of the firmware running on
the SPA.

Check Primary
DNS Server
Setting

160

None

IVR announces the current
setting in the  parameter.

Set Primary
DNS Server

161

Enter IP address
using numbers on
the telephone key
pad. Use the * (star)
key when entering a
decimal point.

Requires password

Check
administration
web server port

170

None

IVR announces the port that
the web server is listening
on. (Default is 80.)

Check LAN IP
Address

210

None

IVR announces the current
IP address of the LAN port.

Check PBX
multicast
address

180

None

IVR announces the current
value.

Set PBX
multicast
address

181

Enter IP address
and port. Use * key
for entering a dot.
For example,
224.168.168.169:80
89 is
224*168*168*169*8
089.

Enter a * between the IP
address and the Port fields.
Requires Password

Enable/Disable
administration
web server

7932

Enter 1 to enable
Enter 0 to disable

Requires password

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IVR Action

IVR
Menu
Choice

Parameters

Notes

Manage the
Auto Attendant
Messages

72255

Enter the message
number, followed by
the pound key. Then
enter 1 to record, 2
to review, 3 to
review, or * to exit.

For more information, see
Chapter 7, “Configuring the
Auto Attendant.”

Manual Reboot
of Unit

732668

None

After you hear “Option
Successful,” hang up. Unit
reboots automatically.

User Factory
Reset of Unit

877778

Enter 1 to confirm
Enter *(star) to
cancel operation

SPA prompts for
confirmation. After
confirming, you hear
“Option Successful.” Hang
up. Unit reboots and all
“User Changeable”
configuration parameters
are reset to factory default
values.

73738

Enter 1 to confirm
Enter * (star) to
cancel operation

SPA prompts for
confirmation. After
confirming, you hear
“Option Successful.” Hang
up. Unit reboots and all
configuration parameters
are reset to factory default
values.

WARNING:
ALL “UserChangeable”
NON-DEFAULT
SETTINGS WILL
BE LOST!
This might
include network
and service
provider data.
Factory Reset of
Unit
WARNING:
ALL NONDEFAULT
SETTINGS WILL
BE LOST!
This includes
network and
service provider
data.

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Using the Interactive Voice Response Unit

2

NOTE The items marked with “Requires Password” only require a password if the

Administrator password is set.

Entering a Password through the IVR
To input the password using the phone keypad, the following translation
conventions apply:

•

To input: A, B, C, a, b, c—press “2’

•

To input: D, E, F, d, e, f—press “3’

•

To input: G, H, I, g, h, i—press “4’

•

To input: J, K, L, j, k, l— press “5’

•

To input: M, N, O, m, n, o—press “6’

•

To input: P, Q, R, S, p, q, r, s—press “7’

•

To input: T, U, V, t, u, v—press “8’

•

To input: W, X, Y, Z, w, x, y, z—press “9’

•

To input all other characters in the Administrator account password, press “0’

For example, to input password test#@1234 by phone keypad, you need to press
the following sequence of digits: 8378001234. This translation convention only
applies to the password input.
STEP 1

After entering a value, press the # (pound) key to indicate end of input.

•

To save value, press 1.

•

To review the value, press 2.

•

To re-enter the value, press 3.

•

To cancel the value entry and return to the main configuration menu, press *’
(star).

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STEP 2

2

•

The final # key is not included in the password value.

•

Saved settings take effect when the telephone is hung-up, and if necessary,
the SPA9000 automatically reboots.

After one minute of inactivity, the unit times out. The user needs to re-enter the
configuration menu from the beginning by pressing * * * *.

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3
Configuring Your System for ITSP
Interoperability

This chapter provides configuration details to help you to ensure that your
infrastructure properly supports voice services.

•

“About the SPA9000 Voice System and SIP,” on page 47

•

“Network Address Translation (NAT) and Voice over IP (VoIP),” on page 49

•

“Firewalls and SIP,” on page 54

•

“Configuring SIP Timer Values,” on page 55

About the SPA9000 Voice System and SIP
The SPA9000 Voice System is implemented using open standards, such as
Session Initiation Protocol (SIP), to help ensure interoperability with all ITSPs that
support SIP. This section provides information about the SIP requests and the
settings that you may need to adjust on your network or your SPA9000 to help
ensure interoperability.
The VoIP telephone service is coordinated by SIP requests and responses,
whether the calls are internal or external. Figure1, “SIP Requests and Responses
for Internal Calls,” on page 48 illustrates the SIP requests and responses between
client stations in the SPA9000 Voice System. The SPA9000 acts as a SIP proxy
and establishes a session. After the session is established, Real Time Protocol
(RTP) traffic flows directly between the two client stations.

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Configuring Your System for ITSP Interoperability
About the SPA9000 Voice System and SIP

Figure 1

SIP Requests and Responses for Internal Calls
SIP UA
2

4

SIP Proxy

RTP
SIP Proxy
3

SIP Proxy
1

SIP UA

Likewise, SIP requests and responses are exchanged to support outbound and
inbound calls that are handled through the ITSP service. In Figure 2, “SPA9000 as a
SIP Proxy for Internet Calls,” UserA and UserB are client stations that are
registered to the SPA9000. When UserA calls UserC, the SPA9000 directs the
request to the SIP proxy at the ITSP, which is then responsible for routing the
request to UserC. After the session is established, RTP is anchored by the
SPA9000.
Figure 2 SPA9000 as a SIP Proxy for Internet Calls
UserC

UserB

UserA

Hub/switch

IP Router (firewall)
Broadband modem
ISP

Internet

Internet (WAN)
Interface

SPA9000
ITSP
SIP Proxy with
media proxy enabled

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Network Address Translation (NAT) and Voice over IP (VoIP)

3

Network Address Translation (NAT) and Voice over IP (VoIP)
NAT is a function that allows multiple devices to share the same public, routable, IP
address to establish connections over the Internet. NAT is present in many
broadband access devices to translate public and private IP addresses. To enable
VoIP to co-exist with NAT, some form of NAT traversal is required.
Some ITSPs provide NAT traversal, but some do not. If your ITSP does not provide
NAT traversal, you have several options.

•

“NAT Mapping with Session Border Controller,” on page 49

•

“NAT Mapping with SIP-ALG Router,” on page 49

•

“Configuring NAT Mapping with a Static IP Address,” on page 49

•

“Configuring NAT Mapping with STUN,” on page 51

NAT Mapping with Session Border Controller
It is strongly recommended that you choose an ITSP that supports NAT mapping
through a Session Border Controller. With NAT mapping provided by the ITSP, you
have more choices in selecting a router.

NAT Mapping with SIP-ALG Router
If the ITSP network does not provide a Session Border Controller functionality, you
can achieve NAT mapping by using a router that has a SIP ALG (Application Layer
Gateway). The WRV200 router is recommended for this purpose, although any
router with a SIP-ALG can be used. By using a SIP-ALG router, you have more
choices in selecting an ITSP.

Configuring NAT Mapping with a Static IP Address
If the ITSP network does not provide a Session Border Controller functionality, and
if other requirements are met, you can configure NAT mapping to ensure
interoperability with the ITSP.

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Configuring Your System for ITSP Interoperability
Network Address Translation (NAT) and Voice over IP (VoIP)

3

Requirements:

•

You must have an external (public) IP address that is static.

•

The NAT mechanism used in the router must be symmetric. See “Determining
Whether the Router Uses Symmetric or Asymmetric NAT,” on page 53.

•

The LAN switch must be configured to enable Spanning Tree Protocol and Port
Fast on the ports to which the SPA devices are connected.

NOTE Use NAT mapping only if the ITSP network does not provide a Session Border

Controller functionality.
STEP 1

Connect to the administration web server, and choose Admin access with
Advanced settings.

STEP 2

Click Voice tab > SIP.

STEP 3

Scroll down to the NAT Support Parameters section, and then enter the following
settings to support static mapping to your public IP address:

•

Handle VIA received, Insert VIA received, Substitute VIA Addr: yes

•

Handle VIA rport, Insert VIA rport, Send Resp To Src Port: yes

•

EXT IP: Enter the public IP address for your router.
Voice tab > SIP: NAT Support Parameters

STEP 4

Click Voice tab > Line N, where N represents the line interface number.

STEP 5

Scroll down to the NAT Settings section.

•

NAT Mapping Enable: Choose YES.

•

NAT Keep Alive Enable: Choose YES (optional).

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Network Address Translation (NAT) and Voice over IP (VoIP)

3

Voice tab > Line N > NAT Settings

STEP 6

Click Submit All Changes.

NOTE You also need to configure the firewall settings on your router to allow SIP

traffic. See “Firewalls and SIP,” on page 54.

Configuring NAT Mapping with STUN
If the ITSP network does not provide a Session Border Controller functionality, and
if other requirements are met, it is possible to use STUN as a mechanism to
discover the NAT mapping. This option is considered a practice of last resort and
should be used only if the other methods are unavailable.
Requirements:

•

STUN is a viable option only if your router uses asymmetric NAT. See
“Determining Whether the Router Uses Symmetric or Asymmetric NAT,” on
page 53.

•

You must have a computer running STUN server software.

•

The LAN switch must be configured to enable Spanning Tree Protocol and Port
Fast on the ports to which the SPA devices are connected.

NOTE Use NAT mapping only if the ITSP network does not provide a Session Border

Controller functionality.
STEP 1

Connect to the administration web server, and choose Admin access with
Advanced settings.

STEP 2

Click Voice tab > SIP.

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Configuring Your System for ITSP Interoperability
Network Address Translation (NAT) and Voice over IP (VoIP)

STEP 3

3

Scroll down to the NAT Support Parameters section, and then enter the following
settings to enable and support the STUN server settings:

•

Handle VIA received: yes

•

Handle VIA rport: yes

•

Insert VIA received: yes

•

Insert VIA rport: yes

•

Substitute VIA Addr: yes

•

Send Resp To Src Port: yes

•

STUN Enable: Choose yes.

•

STUN Server: Enter the IP address for your STUN server.
Voice tab > SIP > NAT Support Parameters

STEP 4

Click Voice tab > Line N, where N is the number of the line interface.

STEP 5

Scroll down to the NAT Settings section.

•

NAT Mapping Enable: Choose yes.

•

NAT Keep Alive Enable: Choose yes (optional).
Voice tab > Line N > NAT Settings

NOTE Your ITSP may require the SPA device to send NAT keep alive messages to

keep the NAT ports open permanently. Check with your ITSP to determine
the requirements.

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Network Address Translation (NAT) and Voice over IP (VoIP)

STEP 6

3

Click Submit All Changes.

NOTE You also need to configure the firewall settings on your router to allow SIP

traffic. See “Firewalls and SIP,” on page 54.

Determining Whether the Router Uses Symmetric or
Asymmetric NAT
STUN does not work on routers with symmetric NAT. With symmetric NAT, IP
addresses are mapped from one internal IP address and port to one external,
routable destination IP address and port. If another packet is sent from the same
source IP address and port to a different destination, then a different IP address
and port number combination is used. This method is restrictive because an
external host can send a packet to a particular port on the internal host only if the
internal host first sent a packet from that port to the external host.

NOTE This procedure assumes that a syslog server is configured and is ready to receive

syslog messages.
STEP 1

Make sure you do not have firewall running on your PC that could block the syslog
port (port 514 by default).

STEP 2

Connect to the administration web server, and choose Admin access with
Advanced settings.

STEP 3

To enable debugging, complete the following tasks:
a. Click Voice tab > System.
b. In the Debug Server field, enter the IP address of your syslog server. This
address and port number must be reachable from the SPA9000.
c. From the Debug level drop-down list, choose 3.

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Firewalls and SIP

STEP 4

3

To collect information about the type of NAT your router is using, complete the
following tasks:
a. Click Voice tab > SIP.
b. Scroll down to the NAT Support Parameters section.
c. From the STUN Test Enable field, choose yes.

STEP 5

To enable SIP signalling, complete the following task:
a. Click Voice tab > Line N, where N represents the line interface number.
b. In the SIP Settings section, choose full from the SIP Debug Option field.

STEP 6

Click Submit All Changes.

STEP 7

View the syslog messages to determine whether your network uses symmetric
NAT. Look for a warning header in the REGISTER messages, such as Warning: 399
spa "Full Cone NAT Detected.”

Firewalls and SIP
To enable SIP requests and responses to be exchanged with the SIP proxy at the
ITSP, you must ensure that your firewall allows both SIP and RTP unimpeded
access to the Internet.

•

•

Make sure that the following ports are not blocked:

•

SIP ports—UDP port 5060 through 5063, which are used for the ITSP line
interfaces

•

RTP ports—16384 to 16482

Also disable SPI (Stateful Packet Inspection) if this function exists on your
firewall.

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Configuring SIP Timer Values

3

Configuring SIP Timer Values
The default timer values should be adequate in most circumstances. However, you
can adjust the SIP timer values as needed to ensure interoperability with your
ISTP. For example, if SIP requests are returned with an “invalid certificate”
message, you may need to enter a longer SIP T1 retry value.
To view the default settings or to make changes, open the Voice > SIP page, and
scroll down to the SIP Timer Values section. .

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4
Configuring Phone Lines and Calling Routing
Behavior

This chapter describes many features that you can configure on the SPA9000 to
ensure smooth handling of all inbound and outbound calls, and ease of use.

•

“Configuring SPA9000 FXS Ports,” on page 57

•

“Configuring Line Interfaces on the SPA9000,” on page 58

•

“Configuring Dial Plans,” on page 66

•

“Managing the Line Selection for Outbound Calls,” on page 78

•

“Managing Caller ID Settings for Outgoing Calls,” on page 82

•

“Call Forwarding Support on SPA9000,” on page 82

•

“Call Transfer Support on SPA9000,” on page 84

•

“Managing Inbound Calls with the Contact List,” on page 85

•

“Managing Inbound Calls with Hunt Groups,” on page 92

•

“Managing Inbound Calls with Shared Line Appearances,” on page 98

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Configuring SPA9000 FXS Ports

4

Configuring SPA9000 FXS Ports
The SPA9000 FXS ports can be used to connect analog phones and fax machines
to the SPA9000 Voice System. A port also can be configured for a Streaming
Audio Server for Music On Hold. See Chapter 6, “Configuring Music on Hold.”

NOTE A fax machine can be connected to the Phone port of the SPA9000. Fax support

through an ITSP line requires a T.38 fax machine on both ends and the availability
of T.38 FAX relay through the ITSP. T.38 support is dependent on fax machine and
network / transport resilience. Linksys makes no guarantee with the use of this
product regarding fax transmission services

STEP 1

Start Internet Explorer, and then enter the IP address of the SPA9000. Click Admin
Login and then click Advanced.

STEP 2

Click Voice tab > FXS N, where N is the port number.

STEP 3

Scroll down to the Subscriber Information section, and then enter the following
settings:

•

Display Name: Enter an extension number of name for the FXS 1 port, such as
Receptionist Area Fax Machine. You can use this extension number to add the
analog phone to the contact list, hunt groups, and shared line appearances.

•

User ID: Enter a three- to four-digit extension number that is not is use by other
extension.

•

If the device is a fax machine, disable echo cancelling. On the FXS N page,
Audio Configuration section, set the FAX Disable ECAN field to yes. Also make
sure that the Preferred Codec is set to G.711u (default setting).

STEP 4

Enter the Dial Plan settings, as needed. See “Configuring Dial Plans,” on page 66.

STEP 5

Click Submit All Changes.

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Configuring Line Interfaces on the SPA9000

4

Configuring Line Interfaces on the SPA9000
You can configure the following types of services on the SPA9000 line interfaces:

•

ITSP service: Up to 16 DID numbers can be supported on each line interface.
You can configure different ITSP accounts on different line interfaces.

•

PSTN service: You can configure a line interface to register the SPA9000 with
a SPA400 to support PSTN lines.

•

SPA400 voice mail service: You can configure a line interface to register the
SPA9000 with a SPA400 to support voice mail server. This SPA400 should
have no more than two PSTN lines connected. If more than two PSTN lines and
voice mail are required, you should reserve one SPA400 exclusively for voice
mail. Exceeding these guidelines will affect the quality of voice mail playback
and command recognition.

•

ISDN services: You can configure a line interface to register the SPA9000 with
a Mediatrix® 4400 ISDN BRI Digital gateway. For more information, refer to the
SPA9000/Mediatrix® 440X ISDN Gateway Configuration Guide. Partners
can find this guide by going to Cisco Partner Central, Voice & Conferencing
page, Technical Resources section. Use the following URL:
www.cisco.com/web/partners/sell/smb/products/
voice_and_conferencing.html#~vc_technical_resources

This section includes the following topics:

•

“Configuring a Line Interface for ITSP Service,” on page 58

•

“Configuring a Line Interface for a SPA400 (PSTN or Voice Mail),” on page 60

•

“Configuring Call Capacity for a Line Interface,” on page 63

Configuring a Line Interface for ITSP Service
STEP 1

Start Internet Explorer, and then enter the IP address of the SPA9000. Click Admin
Login and then click Advanced.

STEP 2

Click Voice tab > Line N, where N represents the line interface number.

STEP 3

From the Line Enable drop-down list, choose yes.

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Configuring Line Interfaces on the SPA9000

STEP 4

Enter the account information for your ITSP account:

•

User ID: The account number or logon name for your ITSP account (often the
same as the phone number)

•

Password: The password for your ITSP account

•

Proxy: The proxy server for your ITSP account
SPA9000 Voice > Line

STEP 5

•

SIP Port: You can keep the default value. Each line must have a unique SIP port
(5060 for Line 1, 5061 for Line 2, 5062 for Line 3, 5064 for Line 4).

•

Contact List: The default value is aa, for the Auto Attendant. As a general
practice, you should leave the default value until after you confirm that the line
is registered. Then you can configure the contact list. For more information, see
“Managing Inbound Calls with the Contact List,” on page 85.

•

Depending on your ITSP network configuration requirements, you may need to
set additional parameters such as Outbound Proxy. Service Provider will
indicate the setting of any additional parameter for each ITSP line.

Click Submit All Changes. The SPA9000 device reboots.

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Configuring Line Interfaces on the SPA9000

STEP 6

4

To verify the registration state, perform the following tasks:

•

After the devices reboot, Click Voice tab > Info. Scroll down to the Line Status
section for the line that you configured (Line 1 Status... Line 4 Status). Verify that
the line is registered. If the line is not registered, you may need to refresh the
browser several times because it can take a few seconds for the registration to
succeed.
SPA9000 Voice > Info > Line Status

•

Use an external phone to place an inbound call to the telephone number that
was assigned by your ITSP. Assuming that you have left the default settings in
place, the Auto Attendant answers the call. You can then dial an extension
number to verify that the call rings to the station.

Configuring a Line Interface for a SPA400 (PSTN or Voice
Mail)
You can configure a line interface to register the SPA9000 to a SPA400 for PSTN
access or voice mail service. To enable the interoperation of the SPA9000 and the
SPA400, you need to enter corresponding information on the SPA9000 Voice >
Line page and on the SPA400 Setup > SPA9000 Interface page. For voice mail
service, also configure the SPA400 Setup > Voicemail Server page.
Before you configure a line interface, be aware of the following factors:

•

The SPA9000 registers to the SPA400. Therefore, the SPA400 must be
connected and available when the SPA9000 attempts to register to it.

•

If you have not yet set a static IP address for the SPA400, you will need to know
the DHCP-obtained IP address. Review the DHCP client list on the router.

NOTE Important: For optimum Voice Mail performance, a SPA400 should be dedicated

to the Voice Mail application when either of the following conditions is met:
1) More than 2 FXO connections are required
—OR—
2) More than 2 users commonly access voice mail at the same time.

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Configuring Line Interfaces on the SPA9000

4

STEP 1

Start Internet Explorer, and then enter the IP address of the SPA9000. Click Admin
Login and then click Advanced.

STEP 2

Click Voice tab > Line N, where N represents the line interface number.

STEP 3

From the Line Enable drop-down list, choose yes.

STEP 4

Enter the following information:

•

User ID: Enter a user ID, such as 9000.
This entry must exactly match the user ID on the SPA400 Setup > SPA9000
Interface page, User ID field. For more information, see “Configuring a SPA400
to Interoperate with the SPA9000,” on page108.

•

Proxy: Enter the IP address of the SPA400.

•

Register Expires: 60
This setting ensures that the SPA9000 and SPA400 are resynchronized every
60 seconds. This setting ensures that any changes in settings are synchronized
on both devices.
SPA9000 Voice > Line: Subscriber Information, Proxy and Registration

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Configuring Line Interfaces on the SPA9000

STEP 5

4

Also in the Proxy and Registration section, enter the following settings to ensure
that calls can be transferred and forwarded to the voice mail server:

•

Set VMSP Bridge to all (required if this line is being used for SPA400 voice mail
service).

STEP 6

STEP 7

•

Set XFER Bridge Mode to all.

•

Set CFWD Bridge Mode to all.

•

SIP Port: You can keep the default value. Each line must have a unique SIP port
(5060 for Line 1, 5061 for Line 2, 5062 for Line 3, 5064 for Line 4).

•

Contact List: The default value is aa, for the Auto Attendant. As a general
practice, you should leave the default value until after you confirm that the line
is registered in the Voice > Info page, Line Status section, Registration State
field. Then you can configure the contact list. For more information, see
“Managing Inbound Calls with the Contact List,” on page 85.

Proceed as needed:

•

If you are using this SPA400 for voice mail service, continue to Step 7.

•

If you are using this SPA400 for PSTN access only, click Submit All Changes to
finish this procedure. You will need to configure the SPA400. For more
information, see “Configuring a SPA400 to Interoperate with the SPA9000,” on
page108 and “Configuring a SPA400 for PSTN Access,” on page111.

Enter the following settings for the SPA400 voice mail service:

•

Mailbox Deposit URL: 900@:5090
The SPA9000 uses this address to deposit voice mail on the voice mail server.

•

Mailbox Manage URL: 800@:5090
The SPA9000 uses this address to access voice mail on the voice mail server.

•

Mailbox Subscribe URL: 8888@:5090
The SPA9000 uses this address to subscribe to voice mail service on the voice
mail server.

•

Mailbox Subscribe Expires: 30
This setting ensures that the SPA9000 and the SPA400 voice mail server are
resynchronized every 30 seconds, and prevents problems when you make
changes in the settings.

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Configuring Phone Lines and Calling Routing Behavior
Configuring Line Interfaces on the SPA9000

SPA9000 Voice > Line

STEP 8

Click Submit All Changes. The SPA9000 device reboots.

NOTE You need to configure the SPA400 with the corresponding settings. See

“Configuring Local Voice Mail Service on a SPA400,” on page113.

Configuring Call Capacity for a Line Interface
Each line interface has a limited number of simultaneous calls that are allowed,
based on the Call Capacity parameter. When the maximum call capacity is
reached, the SPA9000 does not allocate any more calls to that line interface.
This section includes the following topics:

•

“Bandwidth Requirements and Call Capacity,” on page 64

•

“Setting the Call Capacity Parameter,” on page 65

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Configuring Line Interfaces on the SPA9000

Bandwidth Requirements and Call Capacity
The available connection bandwidth determines the maximum number of
simultaneous calls that the system can support with the appropriate audio quality.
Before installing and configuring the Cisco SPA components, use this information
to determine the maximum number of simultaneous VoIP connections that the
system can support. For asymmetric connections, such as ADSL, the maximum
number of calls is determined by the upstream bandwidth. In general it is a good
practice to use no more than 75% of the total available bandwidth for calls. This
provides space for data traffic and helps ensure good voice quality.
The following table provides the approximate bandwidth budget for different
codecs.
Table 1

Bandwidth Budgeting

Codec

Approximate bandwidth budget (kbps)
Each side of
conversation

2 calls

4 calls

6 calls

8 calls

G.711

110

220

440

660

880

G.726-40

87

174

348

522

696

G.726-32

79

158

316

474

632

G.726-24

71

142

284

426

568

G.726-16

63

126

252

378

504

G.729

55

110

220

330

440

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Configuring Line Interfaces on the SPA9000

4

Setting the Call Capacity Parameter
You can set the maximum total number of incoming and outgoing calls on each line
interface. The default value is unlimited. You can set a value from 1 to 15, or leave
the setting as unlimited.

NOTE The SPA9000 does not distinguish between incoming and outgoing calls for call

capacity.

STEP 1

Start Internet Explorer, and then enter the IP address of the SPA9000. Click Admin
Login and then click Advanced.

STEP 2

Click Voice tab > Line N, where N represents the line interface number.

STEP 3

Scroll down to the Subscriber Information section.
From the Call Capacity drop-down list, choose the maximum number of calls to
allow.
SPA9000 Voice > Line: Subscriber Information

STEP 4

Click Submit All Changes.

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Configuring Phone Lines and Calling Routing Behavior
Configuring Dial Plans

4

Configuring Dial Plans
Dial plans determine how the digits are interpreted and transmitted. They also
determine whether the dialed number is accepted or rejected. You can use a dial
plan to facilitate dialing or to block certain types of calls such as long distance or
international.
This section includes information that you need to understand dial plans, as well as
procedures for configuring your own dial plans. This section includes the following
topics:

•

“About Dial Plans,” on page 66

•

“Editing Dial Plans,” on page 74

About Dial Plans
This section provides information to help you understand how dial plans are
implemented.
Refer to the following topics:

•

“Digit Sequences,” on page 66

•

“Digit Sequence Examples,” on page 68

•

“Acceptance and Transmission the Dialed Digits,” on page 71

•

“Dial Plan Timer (Off-Hook Timer),” on page 72

•

“Interdigit Long Timer (Incomplete Entry Timer),” on page 73

•

“Interdigit Short Timer (Complete Entry Timer),” on page 73

Digit Sequences
A dial plan contains a series of digit sequences, separated by the | character. The
entire collection of sequences is enclosed within parentheses. Each digit
sequence within the dial plan consists of a series of elements, which are
individually matched to the keys that the user presses.

NOTE White space is ignored, but may be used for readability.

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Configuring Dial Plans

Digit Sequence

Function

0 1 2 3 4 5 6 7 8 9 0
* #

Enter any of these characters to represent a key
that the user must press on the phone keypad.

x

Enter x to represent any character on the phone
keypad.

[sequence]

Enter characters within square brackets to create
a list of accepted key presses. The user can press
any one of the keys in the list.

.
(period)



•

Numeric range
For example, you would enter [2-9] to allow the
user to press any one digit from 2 through 9.

•

Numeric range with other characters
For example, you would enter [35-8*] to allow
the user to press 3, 5, 6, 7, 8, or *.

Enter a period for element repetition. The dial plan
accepts 0 or more entries of the digit. For
example, 01. allows users to enter 0, 01, 011,
0111, and so on.
Use this format to indicate that certain dialed
digits are replaced by other characters when the
sequence is transmitted. The dialed digits can
be zero or more characters.
EXAMPLE 1: <8:1650>xxxxxxx
When the user presses 8 followed by a sevendigit number, the system automatically replaces
the dialed 8 with 1650. If the user dials
85550112, the system transmits 16505550112.
EXAMPLE 2: <:1>xxxxxxxxxx
In this example, no digits are replaced. When the
user enters a 10-digit string of numbers, the
number 1 is added at the beginning of the
sequence. If the user dials 9725550112, the
system transmits 19725550112

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Configuring Dial Plans

4

Digit Sequence

Function

,

Enter a comma between digits to play an “outside
line” dial tone after a user-entered sequence.

(comma)

EXAMPLE: 9, 1xxxxxxxxxx
An “outside line” dial tone is sounded after the
user presses 9, and the tone continues until the
user presses 1.
!
(exclamation point)

Enter an exclamation point to prohibit a dial
sequence pattern.
EXAMPLE: 1900xxxxxxx!
The system rejects any 11-digit sequence that
begins with 1900.

*xx

Enter an asterisk to allow the user to enter a 2digit star code.

S0 or L0

Enter S0 to reduce the short inter-digit timer to 0
seconds, or enter L0 to reduce the long inter-digit
timer to 0 seconds.

Digit Sequence Examples
The following examples show digit sequences that you can enter in a dial plan.
In a complete dial plan entry, sequences are separated by a pipe character (|), and
the entire set of sequences is enclosed within parentheses.
EXAMPLE: ( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8,
<:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! |
9, 011xxxxxx. | 0 | [49]11 )

•

Extensions on your system
EXAMPLE: ( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8,
<:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx !
| 9, 011xxxxxx. | 0 | [49]11 )
[1-8]xx Allows a user dial any three-digit number that starts with the digits 1
through 8. If your system uses four-digit extensions, you would instead enter
the following string: [1-8]xxx

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Configuring Dial Plans

•

4

Local dialing with seven-digit number
EXAMPLE: ( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8,
<:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx !
| 9, 011xxxxxx. | 0 | [49]111)
9, xxxxxxx After a user presses 9, an external dial tone sounds. The user can
enter any seven-digit number, as in a local call.

•

Local dialing with 3-digit area code and a 7-digit local number
EXAMPLE: ( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8,
<:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx !
| 9, 011xxxxxx. | 0 | [49]11 )
9, <:1>[2-9]xxxxxxxxx This example is useful where a local area code is required.

After a user presses 9, an external dial tone sounds. The user must enter a 10digit number that begins with a digit 2 through 9. The system automatically
inserts the 1 prefix before transmitting the number to the carrier.

•

Local dialing with an automatically inserted 3-digit area code
EXAMPLE: ( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8,
<:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9,
011xxxxxx. | 0 | [49]11 )
8, <:1212>xxxxxxx This is example is useful where a local area code is required

by the carrier but the majority of calls go to one area code. After the user
presses 8, an external dial tone sounds. The user can enter any seven-digit
number. The system automatically inserts the 1 prefix and the 212 area code
before transmitting the number to the carrier.

•

U.S. long distance dialing
EXAMPLE: ( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx |
8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9,
011xxxxxx. | 0 | [49]11 )
9, 1 [2-9] xxxxxxxxx After the user presses 9, an external dial tone sounds. The

user can enter any 11-digit number that starts with 1 and is followed by a digit
2 through 9.

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•

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Blocked number
EXAMPLE: ( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx |
8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! |
9, 011xxxxxx. | 0 | [49]11 )
9, 1 900 xxxxxxx ! This digit sequence is useful if you want to prevent users from
dialing numbers that are associated with high tolls or inappropriate content,
such as 1-900 numbers in the U.S.. After the user press 9, an external dial tone
sounds. If the user enters an 11-digit number that starts with the digits 1900,
the call is rejected.

•

U.S. international dialing
EXAMPLE: ( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx |
8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx
! | 9, 011xxxxxx. | 0 | [49]11 )
9, 011xxxxxx. After the user presses 9, an external dial tone sounds. The user can

enter any number that starts with 011, as in an international call from the U.S.

•

Informational numbers
EXAMPLE: ( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8,
<:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx !
| 9, 011xxxxxx. | 0 | [49]11 )
0 | [49]11 This example includes two digit sequences, separated by the pipe
character. The first sequence allows a user to dial 0 for an operator. The second
sequence allows the user to enter 411 for local information or 911 for
emergency services.

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Configuring Dial Plans

Acceptance and Transmission the Dialed Digits
When a user dials a series of digits, each sequence in the dial plan is tested as a
possible match. The matching sequences form a set of candidate digit sequences.
As more digits are entered by the user, the set of candidates diminishes until only
one or none are valid. When a terminating event occurs, the SPA9000 either
accepts the user-dialed sequence and initiates a call, or else rejects the sequence
as invalid. The user hears the reorder (fast busy) tone if the dialed sequence is
invalid.
The following table explains how terminating events are processed.
Terminating Event

Processing

The dialed digits do not match
any sequence in the dial plan.

The number is rejected.

The dialed digits exactly match
one sequence in the dial plan.

•

If the sequence is allowed by the dial plan, the
number is accepted and is transmitted
according to the dial plan.

•

If the sequence is blocked by the dial plan, the
number is rejected.

A timeout occurs.

The user presses the # key or
the dial softkey on the phone
display.

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The number is rejected if the dialed digits are
not matched to a digit sequence in the dial
plan within the time specified by the
applicable interdigit timer.

•

The Interdigit Long Timer applies when the
dialed digits do not match any digit sequence
in the dial plan. The default value is 10
seconds.

•

The Interdigit Short Timer applies when the
dialed digits match one or more candidate
sequences in the dial plan. The default value is
3 seconds.

•

If the sequence is complete and is allowed by
the dial plan, the number is accepted and is
transmitted according to the dial plan.

•

If the sequence is incomplete or is blocked by
the dial plan, the number is rejected.

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Dial Plan Timer (Off-Hook Timer)
You can think of the Dial Plan Timer as “the off-hook timer.” This timer starts
counting when the phone goes off hook. If no digits are dialed within the specified
number of seconds, the timer expires and the null entry is evaluated. Unless you
have a special dial plan string to allow a null entry, the call is rejected. The default
value is 5.

Syntax for the Dial Plan Timer
SYNTAX: (Ps<:n> | dial plan )

•

s: The number of seconds; if no number is entered after P, the default timer of 5
seconds applies.

•

n: (optional): The number to transmit automatically when the timer expires; you
can enter an extension number or a DID number. No wildcard characters are
allowed because the number will be transmitted as shown. If you omit the
number substitution, <:n>, then the user hears a reorder (fast busy) tone after
the specified number of seconds.

Examples for the Dial Plan Timer

•

Allow more time for users to start dialing after taking a phone off hook.
EXAMPLE: (P9 | (9,8<:1408>[2-9]xxxxxx | 9,8,1[2-9]xxxxxxxxx
| 9,8,011xx. | 9,8,xx.|[1-8]xx)
P9 After taking a phone off hook, a user has 9 seconds to begin dialing. If no
digits are pressed within 9 seconds, the user hears a reorder (fast busy) tone.
By setting a longer timer, you allow more time for users to enter the digits.

•

Create a hotline for all sequences on the System Dial Plan
EXAMPLE: (P9<:23> | (9,8<:1408>[2-9]xxxxxx | 9,8,1[29]xxxxxxxxx | 9,8,011xx. | 9,8,xx.|[1-8]xx)
P9<:23> After taking the phone off hook, a user has 9 seconds to begin dialing. If
no digits are pressed within 9 seconds, the call is transmitted automatically to
extension 23.

•

Create a hotline on a line button for an extension
EXAMPLE: ( P0 <:1000>)
With the timer set to 0 seconds, the call is transmitted automatically to the
specified extension when the phone goes off hook. Enter this sequence in the
Phone Dial Plan for Ext 2 or higher on a client station.

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Interdigit Long Timer (Incomplete Entry Timer)
You can think of this timer as the “incomplete entry” timer. This timer measures the
interval between dialed digits. It applies as long as the dialed digits do not match
any digit sequences in the dial plan. Unless the user enters another digit within the
specified number of seconds, the entry is evaluated as incomplete, and the call is
rejected. The default value is 10 seconds.

NOTE This section explains how to edit a timer as part of a dial plan. Alternatively, you can

modify the Control Timer that controls the default interdigit timers for all calls. See
“Resetting the Control Timers,” on page 77.

Syntax for the Interdigit Long Timer
SYNTAX: L:s, ( dial plan )

•

s: The number of seconds; if no number is entered after L:, the default timer of
5 seconds applies.

•

Note that the timer sequence appears to the left of the initial parenthesis for the
dial plan.

Example for the Interdigit Long Timer
EXAMPLE: L:15, (9,8<:1408>[2-9]xxxxxx | 9,8,1[2-9]xxxxxxxxx |
9,8,011xx. | 9,8,xx.|[1-8]xx)
L:15, This dial plan allows the user to pause for up to 15 seconds between digits

before the Interdigit Long Timer expires. This setting is especially helpful to users
such as sales people, who are reading the numbers from business cards and other
printed materials while dialing.

Interdigit Short Timer (Complete Entry Timer)
You can think of this timer as the “complete entry” timer. This timer measures the
interval between dialed digits. It applies when the dialed digits match at least one
digit sequence in the dial plan. Unless the user enters another digit within the
specified number of seconds, the entry is evaluated. If it is valid, the call proceeds.
If it is invalid, the call is rejected. The default value is 3 seconds.

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Syntax for the Interdigit Short Timer

•

SYNTAX 1: S:s, ( dial plan )
Use this syntax to apply the new setting to the entire dial plan within the
parentheses.

•

SYNTAX 2: sequence Ss
Use this syntax to apply the new setting to a particular dialing sequence.
s: The number of seconds; if no number is entered after S, the default timer of 5
seconds applies.

Examples for the Interdigit Short Timer

•

Set the timer for the entire dial plan.
EXAMPLE: S:6, (9,8<:1408>[2-9]xxxxxx | 9,8,1[2-9]xxxxxxxxx |
9,8,011xx. | 9,8,xx.|[1-8]xx)
S:6, While entering a number with the phone off hook, a user can pause for up

to 15 seconds between digits before the Interdigit Short Timer expires. This
setting is especially helpful to users such as sales people, who are reading the
numbers from business cards and other printed materials while dialing.

•

Set an instant timer for a particular sequence within the dial plan.
EXAMPLE: (9,8<:1408>[2-9]xxxxxx | 9,8,1[2-9]xxxxxxxxxS0 |
9,8,011xx. | 9,8,xx.|[1-8]xx)
9,8,1[2-9]xxxxxxxxxS0 With the timer set to 0, the call is transmitted automatically

when the user dials the final digit in the sequence.

Editing Dial Plans
You can edit dial plans and can modify the control timers.

Editing the System Dial Plan
Follow this procedure to edit the system dial plan, which will be auto-provisioned
to the first extension of each phone.
STEP 1

Connect to the SPA9000 administration web server, and choose Admin access
with Advanced settings.

STEP 2

Click Voice tab > SIP.

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STEP 3

4

Scroll down to the PBX Phone Parameters section.
SPA9000 Voice tab > SIP page: PBX Phone Parameters section

STEP 4

Enter the digit sequences in the Dial Plan field. For more information and examples,
see “Digit Sequences,” on page 66.

NOTE Separate each digit sequence with a pipe character, and enclose the entire

set of digit sequences within parentheses. Refer to the following example:
(9,8<:1408>[2-9]xxxxxx | 9,8,1[2-9]xxxxxxxxx |
9,8,011xx. | 9,8,xx.|[1-8]xx)
STEP 5

Click Submit All Changes. The phones reboot.

STEP 6

Verify that you can successfully complete a call using each digit sequence that
you entered in the dial plan.

NOTE If you hear a reorder (fast busy) tone, you need to review your entries and

modify the dial plan appropriately. See “Digit Sequences,” on page 66.

Entering a Phone Dial Plan
The phone dial plan is automatically updated when the system phone dial plan is
modified. There are special cases where it is required to enter the phone dial plan
directly on the phone. Follow the procedure below to enter a dial plan for a
particular extension on a client station in the following cases:

•

The phone is outside the SPA9000 multicast domain (e.g. when the phone is
connected to the SPA9000 via a VPN connection).

•

For additional extensions on an existing phone (the system wide phone dial
plan is propagated only to extension 1 of all phones).

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STEP 1

Start Internet Explorer, and then enter the IP address of the SPA9000. Click Admin
Login and then click Advanced.

STEP 2

Click the PBX Status link near the top right corner or lower left corner of the page,
to view the list of client stations.

STEP 3

Click the hyperlink in the IP Address column for the first phone that you want to
configure. The telephone configuration page appears in a separate browser
window.

STEP 4

Click the Ext 1 tab, or the tab for the extension that you want to configure.

STEP 5

Scroll down to the Dial Plan section.

STEP 6

Enter the digit sequences in the Dial Plan field.

•

The default (US-based) system-wide dial plan appears automatically in the
field. You can delete digit sequences, add digit sequences, or replace the
entire dial plan with a new dial plan. For more information and examples, see
“Digit Sequences,” on page 66.

•

Separate each digit sequence with a pipe character, and enclose the entire set
of digit sequences within parentheses. Refer to the following example:
(9,8<:1408>[2-9]xxxxxx | 9,8,1[2-9]xxxxxxxxx | 9,8,011xx.
| 9,8,xx.|[1-8]xx)

STEP 7

Click Submit All Changes. The phone reboots.

STEP 8

If you need to configure a dial plan for any other extensions on the phone
(depending on the model), click the appropriate Extension tab, enter the dial plan,
and submit the changes.

STEP 9

Click the browser’s Back button to return to the list of phones.

STEP 10 Repeat this procedure for each client station that needs a unique dial plan.
STEP 11 Verify that you can successfully complete a call using each digit sequence that

you entered in the dial plan.

NOTE If you hear a reorder (fast busy) tone, you need to review your entries and modify

the dial plan appropriately. See “Digit Sequences,” on page 66.

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Entering the Line Interface Dial Plan
This dial plan is used to strip steering digits from a dialed number before it is
transmitted out to the carrier.
STEP 1

Connect to the administration web server, and choose Admin access with
Advanced settings.

STEP 2

Click Voice tab > Line N, where N represents the line interface number.

STEP 3

Scroll down to the Dial Plan section.

STEP 4

Enter the digit sequences in the Dial Plan field. For more information, see “About
Dial Plans,” on page 66.

STEP 5

Click Submit All Changes.

Resetting the Control Timers
You can use the following procedure to reset the default timer settings for all calls.

NOTE If you need to edit a timer setting only for a particular digit sequence or type of call,

you can edit the dial plan. See “About Dial Plans,” on page 66.

STEP 1

Connect to the administration web server, and choose Admin access with
Advanced settings.

STEP 2

Click Voice tab > Regional.

STEP 3

Scroll down to the Control Timer Values section.

STEP 4

Enter the desired values in the Interdigit Long Timer field and the Interdigit Short
Timer field. Refer to the definitions at the beginning of this section.

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Managing the Line Selection for Outbound Calls

4

Managing the Line Selection for Outbound Calls
When a user places an outbound call, the SPA9000 chooses a line based on the
group membership of the station and the Call Routing Rule for the line interface.
You can configure the settings to route calls through particular lines, based on
factors such as the dialed number, the destination, or the corporate identity that
you want to represent.
This feature can be used whether the SPA9000 line interface is configured for an
ITSP or a SPA400 voice gateway.
This section includes the following topics:

•

“Line Availability,” on page 78

•

“Configuring a Call Routing Rule,” on page 79

•

“Entering a Call Routing Rule,” on page 81

Line Availability
The SPA9000 considers a line to be available for an outgoing call if the following
conditions are met:

•

The line is enabled in the configuration and is functioning. See “Configuring
Line Interfaces on the SPA9000,” on page 58.

•

The line is authorized by the Call Routing Rule for the dialed number.
See“Configuring a Call Routing Rule,” on page 79.

•

The line has capacity to take more calls. See “Configuring Call Capacity for a
Line Interface,” on page 63.

•

The Dial Plan for this line allows the dialed number. See “Configuring Dial
Plans,” on page 66.

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Configuring a Call Routing Rule
A Call Routing Rule is a special dial plan that specifies the lines that can be used to
transmit a dialed number. The same number pattern can apply to more than one
line.
SYNTAX: ( <:Lw,x,y,z>number-pattern | <:Lw,x,y,z>number-pattern |
<:Lw,x,y,z>number-pattern | <:Lw,x,y,z>number-pattern )

•

L: A signifier for “Line”

•

w, x, y, z: The number of the line interface (L1 for Line 1, L2 for Line 2, and so on)

•

port: The port number

NOTE This parameter applies only if the line interface is configured for a SPA400.

•

number-pattern: The dialed sequence that can use the specified lines

•

Other elements:

•

Enclose the entire call routing rule in parentheses.

•

Use a comma to separate each digit sequence within a number-pattern.

•

Use a pipe character (|) to separate each call routing rule.

NOTE The Call Routing rules use the same digit sequences as the Dial Plan rules. See

“Digit Sequences,” on page 66.

•

Allowing any line interface to be used
EXAMPLE: (<:L1,2,3,4>9xx.)
Any line interface can be used for any dialed sequence that begins with 9 and
includes at least two additional numbers. The SPA9000 chooses an available
line from the list, proceeding in the listed order. See “Line Availability,” on
page 78.

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•

4

Different line interfaces for U.S., international, and 1-800 numbers
EXAMPLE: (<:L1,2>9xx. | <:L3>011852xx. | <:L4>1800xxxxxxx)
This example has three parts:

•

•

<:L1,2>9xx. Line 1 and Line 2 can be used if the sequence starts with 9
and includes at least two additional digits.

•

<:L3>011852xx. Line 3 can be used if the sequence starts with 011852
and includes at least two additional digits.

•

<:L4>1800xxxxxxx. Line 4 can be used if the sequence starts with 1800
and includes at least seven additional digits.

Separate lines for long distance and local calling
EXAMPLE: (<:L1>9xx. | <:L2>8xx.)
Line 1 is used for any dialed sequence that starts with 9. Line 2 is used for any
dialed sequence that starts with 8. This call routing rule is effective if users
understand that 9 is the steering digit for long distance (with Line 1 configured
for the ITSP and its inexpensive long distance service) and that 8 is the steering
digit for local calls (with Line 2 configured for a SPA400 that has local phone
lines connected).

•

Specifying a hunt order for FXO lines on one SPA400 unit
EXAMPLE: (<:L2{1,2}>8xx)
Line 2 is configured for a SPA400. This line is used for any dialed sequence that
starts with 8. The preference is to seize port 1. If port 1 is unavailable, the next
preference is to seize line 2. If neither port is available, then the call fails.

•

Specifying a hunt order for FXO lines on multiple SPA400 units
EXAMPLE: (<:L4{1},3,2{2,4}>9xx)
Lines 2, 3, and 4 are configured for SPA400 units. These lines are used for any
dialed sequence that starts with 9. The preference is to use Line 4, port 1. If it is
unavailable, then any port on Line 3 can be used. If Line 3 is unavailable, then
the next choice is Line 2, port 2. If it is unavailable, the next choice is Line 2, port
4. If it is unavailable, then the call fails.

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•

4

Wildcard characters
EXAMPLE: (<:L1>51*,577?)
In this example, Line 1 can be used when either of the following number
patterns is dialed:

•

The user dials a number that starts with 51, followed by any other
characters.

•

The user dials any four-digit number starting with 577.

Entering a Call Routing Rule
Use the following procedure to enter a call routing rule.
STEP 1

Start Internet Explorer, and then enter the IP address of the SPA9000. Click Admin
Login and then click Advanced.

STEP 2

Click Voice tab > SIP.

STEP 3

Scroll down to the PBX Parameters section.

STEP 4

In the Call Routing Rule field, enter the rule that you want to apply

STEP 5

Click Submit All Changes.

STEP 6

To verify your progress, place a call to a phone that has caller ID, and confirm that
the expected number appears.

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Managing Caller ID Settings for Outgoing Calls
By default, outgoing calls through an ITSP line are identified by the User ID (usually
the phone number) and Display Name of the selected line interface. Alternatively,
you can map a DID number to a phone extension so that all outbound calls from
that extension will identify the caller by the DID number and the assigned phone
display name. (See “Supporting Multiple DID Numbers Per Line Interface,” on
page 87.)

NOTE Caller ID for a PSTN line is controlled by the phone company. Caller ID through an

ITSP line should work as described, but if not, contact your ITSP to see what is
allowed for caller ID configuration.

STEP 1

Start Internet Explorer, and then enter the IP address of the SPA9000. Click Admin
Login and then click Advanced.

STEP 2

Click Voice > Line N, where N represents the line interface number.

STEP 3

Scroll down to the Subscriber Information section.

STEP 4

Enter the desired display name for Caller ID in the Display Name field.

STEP 5

Click Submit All Changes.

Call Forwarding Support on SPA9000
SPA9000 supports the following Call Forward settings on the client station:

•

CFWD All: Forwards all calls unconditionally

•

CFWD Busy: Forwards calls received when the station is busy

•

CFWD NoAns: Forwards calls when there is no answer

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Call Forwarding Support on SPA9000

SPA9000 supports the call forward scenarios listed in the following table.

Called
Party

Caller

Forward Target

Remarks

Client
station

Client
station

Client station

Proxy only; no direct
involvement

Client
station

Client
station

External

Very similar to calling external
number

Client
station

External

Client station

ACKs the 302 from called party.
Then INVITEs the target.
NOTE If the original INVITE is
forked to more than one client
station, call forward is not
performed.

Client
station

External

External

ACKs the 302 from called party.
Sends 200 to caller, then blind
REFERs caller to target.
NOTE If the original INVITE is
forked to more than one client
station, call forward is not performed

External

Client
station

External

ACKs the 302 from called party,
then INVITEs the target

NOTE When an incoming call from the ITSP is forked to multiple client stations, the

SPA9000 does not honor the 3xx response returned by any of the client stations. If
you wish the call to be forwarded to voicemail or another client station when it is not
picked up, you can specify the optional “cfwd=target” syntax in the Contact List.

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Call Transfer Support on SPA9000

4

Call Transfer Support on SPA9000
You can configure the bridge mode for call forward and call transfer.

Call Forward Bridge Mode
The normal way of performing the call forwarding operation is for the SPA9000 to
send a (blind) SIP REFER to the calling device to let it contact the target number
directly. It then drops out of the call completely. This requires the calling device to
understand the SIP signaling involved and the operation permitted by the
underlying service provider. The SPA400 cannot handle this operation.
With bridging, the SPA9000 maintains two separate call legs throughout the call:
one with the caller and one with the call forward target. The two call peers
connect only with the SPA9000, while the SPA9000 acts as a proxy for the RTP
packets exchanged between the two parties. On the Voice > Line N page, Proxy
and Registration section, the CWFD Bridge Mode field has two possible values:

•

none—Do not bridge forwarded calls (use the normal REFER method)

•

all—Bridge all forwarded calls

Call Transfer Bridge Mode
The normal way of performing this operation is for the SPA9000 to send a SIP
REFER method to the calling device to let it contact the transfer target directly. The
SPA9000 then drops out of the call completely. This requires the calling device
(the transferee) and the target device to understand the SIP signaling involved and
the operation permitted by the underlying service providers. Note that the call
legs with transferee and the transfer target might be with different ITSP. The
SPA400, for instance, cannot handle this operation.
With bridging, the SPA9000 maintains two separate call legs throughout the call:
one with the transferred call and one with the transfer target. The two call peers
connect only with the SPA9000, while the SPA9000 acts as a proxy for the RTP
packets exchanged between the two parties. On the Voice > Line N page, Proxy
and Registration section, the XFER Bridge Mode field has three possible values:

•

none: Do not bridge call transfer (use the normal REFER method)

•

all: Bridge all call transfer

•

all except same line: Bridge call transfer only between different line interfaces

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Managing Inbound Calls with the Contact List

4

Managing Inbound Calls with the Contact List
You can use the Contact List to route inbound calls to the Auto Attendant, to a
receptionist, to a client station, to a group of stations, or to a combination of these.

•

“Routing an Inbound Call to the Auto Attendant,” on page 85

•

“Routing an Inbound Call to a Receptionist or Client Stations,” on page 85

•

“Example Contact List Rules,” on page 86

•

“Entering a Contact List Rule,” on page 91

Routing an Inbound Call to the Auto Attendant
By default, all inbound calls are routed to the Auto Attendant (aa). This automated
system answers inbound calls by playing pre-recorded voice message that asks
the caller to enter the desired extension. If you want only the Auto Attendant to
receive a call, keep the default setting, aa, in the Contact List field on the Voice >
Line N page, Subscriber Information section, for each line interface. For more
information, see Chapter 7, “Configuring the Auto Attendant.”

Routing an Inbound Call to a Receptionist or Client Stations
You can route an inbound call to a receptionist or to client stations by using a
Contact List. You specify the Contact List for each line interface (Line 1, Line 2, Line
3, Line 4). For example, if Line1 is configured for an ITSP account, and a call is
placed to a Direct Inward Dialing (DID) number for that account, then the call is
routed to the Contact List that is specified on the Line 1 configuration page.
Likewise, if Line 2 is configured for a SPA400 that has PSTN lines attached, and a
call is placed to the associated PSTN phone number, then the call is routed as
specified in the Voice > Line page, Subscriber Information section, Contact List
field.

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Example Contact List Rules
The following examples show rules that you can enter to route incoming calls.

NOTE The SPA9000 alerts all registered clients stations if * is used in the Contact List

(SPA9000 Voice > Line N page > Subscriber Information section).

•

Routing calls to a receptionist
EXAMPLE: 100
An incoming call to any DID number on this line interface causes station 100 to
ring. The receptionist answers the call. If the call is not answered, it
automatically goes to the voice mailbox for station 100, assuming that voice
mail is configured.

•

Routing calls simultaneously to two or more stations
EXAMPLE: 100, 104
An incoming call to any DID number on this line interface causes station 100
and station 104 to ring. Either station can answer the call.

NOTE The list of extension numbers may include * to represent multiple wildcard

characters or ? to represent a single wildcard character. For example, 10?
represents all stations numbered 100 through 109.

•

Special routing for different DID numbers
EXAMPLE: 9725550155:100|9725550156:101, 102
An incoming call to 972-555-0155 causes station 100 to ring. An incoming call
to 972-555-0156 causes station 101 and station 102 to ring simultaneously.

NOTE In this example, the rules are separated by a pipe character (|) to indicate an

“or” condition.

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•

4

Routing calls to a station and forwarding unanswered calls to voice mail
EXAMPLE 1: 5300, cfwd=vm25300
An incoming call through this line interface causes station 5300 to ring. If there
is no answer, the call is forwarded to the voice mail server on line interface 2,
mailbox number 5300. The time interval is determined by the value Cfwd No
Ans Delay field, which is located below the Contact List field on the Voice >
Line page. The default value is 20 seconds.
EXAMPLE 2: 4085550122:5001|4085550123:5000,cfwd=aa
An incoming call to 408-555-0122 causes station 5001 to ring. An incoming call
to 408-555-0123 causes station 5000 to ring. If station 5000 does not answer
its call, the call is forwarded to the Auto Attendant. The time interval is
determined by the value Cfwd No Ans Delay field, which is located below the
Contact List field. The default value is 20 seconds.

•

Routing a call with a hunt rule
EXAMPLE: 530?,hunt=ra;10;2,cfwd=vm25404
An incoming call through this line interface causes one station in the group
5300 through 5309 to ring. The station is chosen randomly (ra). After 10
seconds, if the call is unanswered, then another station is chosen randomly
from the remaining stations. The system cycles through the list two times. If the
call is unanswered, it is forwarded to the voice mail server on line interface 2,
mailbox 5404.

NOTE A hunt rule in the contact list applies only to calls on the selected line

interface. You also can create hunt groups that apply to all lines. For more
information and additional examples of syntax that can be used in a hunt rule
in the Contact List, see “Managing Inbound Calls with Hunt Groups,” on
page 92.

Supporting Multiple DID Numbers Per Line Interface
An ITSP can provide a block of DID numbers, for example with a main number of
4085553000, and additional DID numbers from 4085553001–4084443009. The
ITSP can identify the local client stations to which an external incoming call should
be routed. Linksys recommends including this information in the TO header of the
incoming INVITE while the request-URI is addressed to the line interface user-id. In
the INVITE, the ITSP indicates the DID number in the TO header user-id field.

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EXAMPLE SIP Header 1:
INVITE sip:4089993000@itsp1.com SIP/2.0
To: 
Alternatively, the DID number can be indicated as a parameter in the TO header
with a configurable parameter name, such as didn.
EXAMPLE SIP Header 2:
INVITE sip:4089993000@itsp1.com SIP/2.0
To: ;didn=4089993003
You can identify the field to use for the DID number and the parameter name on the
Voice > SIP page, PBX Parameters section, SIP DIDN and SIP DIDN Param
Name fields. For the first example above, these two fields are ignored; for the
second example, SIP DIDN is set to TO Param and SIP DIDN Param Name is set to
didn.
The Contact List is used to route the calls to a client station based on DID numbers
that are embedded in the INVITE message.
EXAMPLE Contact List Rule:
4089993000:aa|4089993001:3001|4089993002:3002|…|4089993009:3
009
An incoming call to the main number is answered by the Auto-Attendant, while
calls to the other nine DID numbers are routed to dedicated private extensions.

Supporting Direct Inward Dialing to Phone Extensions
Direct Inward Dialing allows the external users to dial directly any phone extension
in the SPA9000 Voice System, without passing through the Auto Attendant or the
receptionist.
Before proceeding with the configuration you need to have the full
correspondence between the external (DID) number and the extension number.
Table 1, “DID-to-Extension Mapping Example,” on page 89 provides an example.

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Table 1 DID-to-Extension Mapping Example

DID number

Extension number

User

408-555-5550

aa

Auto Attendant

408-555-5551

101

User 1

408-555-5552

102

User 2

408-555-5553

103

User 3

408-555-5554

104

User 4

408-555-5650

500

Support (hunt) group

Be aware of the following factors:

•

Direct Inward Dialing requires network support for SIP trunking DID.

•

It is important that the DID number format match exactly the format of the
number signaled in the SIP trunk.

•

Please check with your Service Provider to confirm the availability of this
feature and the correct DID number format, before proceeding with this
configuration.

SYNTAX:
:+|:+|:+|:+
|:
+|:+|:+|:+|
:+|
:+|:+|:+|:+|
:+|:+|:+|

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EXAMPLE:
4085555550:aa|4085555551:+101|4085555552:+102|4085555553:+10
3|4085555554:+104|4085555650:+500|

•

Enter a plus (+) or a minus (-) before the extension number to achieve the
desired results for caller ID and call routing:

•

+  The extension will be alerted when there is an incoming call to
the DID number, and the DID number will be used as the local user-ID in
outbound SIP requests, along with the display-name of the extension.

•

-  The extension will not be alerted when there is an incoming call
to the DID number. However, the DID number will be used as the local userID in outbound SIP requests, along with the display-name of the extension.
DIDn#: DID number, matching the format of the number signaled in the SIP
trunk

•

Extn#: Target extension number or a for Auto Attendant

•

Default route: The default route (extension, hunt group or Auto Attendant) to be
used in case the incoming target number does not match any of the DID
numbers in the contact list with the format described in the sections above.

•

Call forwarding: Optionally, add call forward information to the Contact List to
specify how unanswered calls are handled.
It is useful to add call forward information to ensure that calls are answered
when the designated phone is unstaffed, as may be the case during lunch time
or after hours. In this case, if the call is not answered within a specified time, the
call is routed to another phone or to the Auto Attendant.
SYNTAX:
:+,cfwd=|:+,cfwd=|:+,cfwd=
EXAMPLE:
4085555550:aa|4085555551:+101,cfwd=aa|4085555552:+102|4085
555553:+103|4085555554:+104|4085555650:+500|

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Entering a Contact List Rule
Use the following procedure to enter a contact list rule.
STEP 1

Connect to the SPA9000 administration web server, and choose Admin access
with Advanced settings.

STEP 2

Click Voice tab > Line N, where N represents the line interface number.

STEP 3

Scroll down to the Subscriber Information section.
SPA9000 Voice > Line: Subscriber Information

STEP 4

Enter the desired rules in the Contact List field. For more information, see “Example
Contact List Rules,” on page 86 and “Entering a Contact List Rule,” on page 91.

NOTE The maximum length of the  parameter is 383 characters.

STEP 5

If you included a call forward rule, modify the Cfwd No Ans Delay parameter to
specify the number of seconds that elapse before a call is considered to be
unanswered.

STEP 6

Click Submit All Changes. The SPA9000 reboots.

STEP 7

To verify your progress, make an inbound call and verify that the call is handled as
you intended.

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Managing Inbound Calls with Hunt Groups
A Hunt Group is a group of client stations that are treated as one extension for the
purpose of managing inbound calls. The client stations in the hunt group can be
alerted simultaneously or sequentially. The first client station to pick up the call
establishes a private connection to the caller.

NOTE Compare a Shared Line Appearance to a Hunt Group. A Hunt Group may ring

several stations simultaneously, but only one station can answer and manage the
call. A Shared Line Appearance also rings multiple lines simultaneously but allows
multiple stations to share and manage the call. For more information, see “Managing
Inbound Calls with Shared Line Appearances,” on page 98.
There are two types of hunt rules:

•

Global Hunt Rule: The hunt rule applies to all line interfaces. Enter this type
of hunt rule on the Voice tab > SIP page > PBX Parameter section, Hunt
Groups field.

•

Line-Specific (Contact List) Hunt Rule: The hunt rule applies to a
particular line interface. Enter this type of hunt rule on the Voice tab > Line N
page > Subscriber Information section > Contact List field. For example, if
you enter this rule on the Line 1 page, it applies only to the calls that are
received on Line 1.

The syntax for the two types of rules varies, as described later in this section. If
you define a hunt group both in the Contact List and on the SIP page > PBX
Parameters section, the call forward parameter from the Contact List is ignored;
instead, the call forward parameter defined on the SIP page > PBX Parameter
section is used.

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Syntax for Hunt Rules
•

Global Hunt Rule
(SIP page > PBX Parameters section > Hunt Groups field)
SYNTAX: extension:[name=”name”,]station[,station[,station
[…]]],hunt=hrule[,cfwd=target]

•

Line-Specific Hunt Rule
(Line N page > Subscriber Information section > Contact List field)
SYNTAX: station[,station[,station[…]]],hunt=hrule;
[,cfwd=target]

Parameters

•

extension: An extension number for the hunt group

•

name: A name for the hunt group, which will appear in the Corporate
Directory

•

station: The extension numbers; the wildcard symbols ? and * can be
used to represent one or more characters

•

hunt=hrule: The hunt order, ring interval, and maximum duration, in the
following format: hunt=algo;interval;max

-

-

algo: The hunt order.
–

re: Restart. Hunting starts at the beginning of the list. If the first station
does not answer within the specified interval (see below), the hunt
proceeds through the stations in sequential order.

–

ne: Next. The system determines the station that was chosen in the
previous hunt, and hunting starts with the next station in the list. If that
station does not answer within the specified interval (see below), the
hunt proceeds through the stations in sequential order.

–

ra: Random order. The system randomly chooses a station from the list. If
the selected station does not answer within the specified interval (see
below), the hunt proceeds randomly through the unchosen stations until
each station is tried.

–

al: All. The system rings all the stations at the same time.

interval: The number of seconds to wait for one station to answer,
before choosing another station. If interval is *, the hunt is stopped at the
first station that starts ringing, and rings the station until it answers, or the
caller hangs up, or the station's ringer times out.

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•

4

max: The maximum duration of the hunt, either in seconds or cycles.
When this limit is reached, the call is rejected or is forwarded to the
specified call forward number (see below).
–

If max is greater than interval, it represents the total time in seconds to
hunt.

–

If max is less than interval, it represents the maximum number of times
to cycle through the hunt group. If max is 0, hunting continues indefinitely
until the caller either hangs up or the call is answered. Exceptions: This value
is ignored if algo = al, or interval = * (but it must be present and
should be set to 1).

cfwd=target: If the call is unanswered and the maximum hunting duration
has been met, the call is forwarded to the specified number. When
forwarding the call, the device sends a 302 response to the ITSP.

NOTE

- The call forward settings for the individual stations are ignored
during hunting. Instead, the call forward settings in the Contact List are
used.
- You cannot forward from one hunt group to another hunt group.

Examples for Hunt Rules
•

Contacting a group of stations simultaneously
Global Rule: 500:name=”Sales”,101,102,103,hunt=al
Line-Specific Rule: 101,102,103,hunt=al
For the global rule, this hunt group is assigned an extension number (500) and
is given a name, Sales. For both examples, the rule includes three stations
(101,102,103). An incoming call rings all three stations simultaneously
(hunt=al).

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Using wildcard characters to specify a range of station numbers
Global Rule: 500:name=”TechSupport”,1*,hunt=al
Line-Specific Rule: 1*,hunt=al
In this example, the * symbol is a wildcard character that represents any
number of digits. An incoming call simultaneously rings all stations that begin
with the number 1.

•

Creating multiple hunt groups with the pipe character
Global Rule: 500:name=”TechSupport”,101,102,103,hunt=al|
600:name=”AccountSupport”,2*,hunt=al
Line-Specific Rule: 101,102,103,hunt=al | 2*,hunt=al
Two hunt groups are created. One hunt rule applies to stations 101, 102, and
103. The other hunt rule applies to all stations that begin with the number 2.

•

Ringing stations sequentially with call forwarding to the Auto Attendant
Global Rule: 300:name=”TechSupport”,101,102,100,hunt=re;
20;2,cfwd=aa
Line-Specific Rule: 101,102,100,hunt=re;20;2,cfwd=aa
An incoming call is routed sequentially from the beginning of the list (hunt=re).
Every new call is routed to the first station, 101. If the call is unanswered, it
cycles to station 102 and then to station 100. Each phone rings for 20 seconds
(20). After 2 cycles, an unanswered call is forwarded to the Auto Attendant
(cfwd=aa).

•

Ringing stations randomly with call forwarding to voice mail
Global Rule: 400:name=”Sales”,101,102,103,hunt=ra;30;1,
cfwd=vm2100
Line-Specific Rule: 101,102,103,hunt=ra;30;1,cfwd=vm2100
An incoming call is routed in random order (hunt=ra). A new call is routed to a
randomly chosen station. If the call is unanswered, it cycles through the
stations in random order. Each phone rings for 30 seconds (30). After 1 cycle is
completed, an unanswered call is forwarded to the voice mail for station 100,
where Line 2 is configured for the voice mail service (1,cfwd=vm2100).

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Resuming from a previous hunt with call forwarding to a receptionist
Global Rule: 500:name=”Scheduling”,102,103,101,hunt=ne;
45;240,cfwd=100
Line-Specific Rule: 102,103,101,hunt=ne; 45;240,cfwd=100
An incoming call is routed in “next station” order (hunt=ne). When a new call is
received, the system resumes the previous hunt. For example, suppose that
station 103 answered the most recent call to the group. A new call is routed to
the next station in the list, which is station 101. If the call is unanswered, then it
cycles to station 102, and so on. Each phone rings for 45 seconds (45;). After a
total of 240 seconds has elapsed, the call is forwarded to station 100
(240,cfwd=100).

Creating a Hunt Rule
Use the following procedure to create a global hunt rule or a line-specific hunt rule.

NOTE For information about creating a hunt rule within a Contact list, see Managing

Inbound Calls with the Contact List, page 85.

STEP 1

Start Internet Explorer, and then enter the IP address of the SPA9000. Click Admin
Login and then click Advanced.

STEP 2

Proceed as needed, depending on the type of hunt rule:

•

Global Rule: Click Voice tab > SIP. Scroll down to the PBX Parameters
section. Find the Hunt Groups field.

•

Line-Specific Rule: Click Voice tab > Line N, where N is the line interface
number. Scroll down to the Subscriber Information section. Find the
Contact List field.

STEP 3

Type your hunt group rule. See “Syntax for Hunt Rules,” on page 93 and “Examples
for Hunt Rules,” on page 94.

STEP 4

Click Submit All Changes. The phones reboot.

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STEP 5

4

To verify the settings:

•

Global Rule: Place a call to the new hunt group by entering the extension
number or by using the Corporate Directory to select the group name and
dial the group.

•

Line-Specific Rule: From an external phone, such as a cellular phone, place
a call to the phone number that is associated with the line interface. Verify
that the call is routed according to the specified settings.

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Managing Inbound Calls with Shared Line Appearances
A shared line appearance (SLA) allows multiple stations to share an extension
number and to manage a call as a group.

About Shared Line Appearances
An incoming call to an SLA causes all stations to ring simultaneously. The phones
display the line status, such as idle, ringing, or busy.
Be aware of the following factors:

•

If a shared line is being used by one of the stations in the group, no one can use
the corresponding line key until the line is released. A message appears on the
phone display to indicate that a shared call is active.

•

All stations with a line key for the SLA can monitor the status based on the
appearance of the line key button:

•

Green: The SLA is available.

•

Flashing Red Quickly: A call is ringing the SLA and has not been answered.

•

Flashing Red Intermittently: A call is active at another station.

•

Flashing Red Slowly: A call is on hold.

•

Solid Red: A call is active on this station.

•

All stations with a display can monitor the status of any SLA that is configured.
The icon next to the line key will change to show ringing, off-hook, or busy "<->" until the phone is back on-hook.

•

Comparing the SLA to a Hunt Group, both may ring several stations
simultaneously. However, only one station in a Hunt Group can answer and
manage the call. For more information, see “Managing Inbound Calls with Hunt
Groups,” on page 92.

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STEP 1

Start Internet Explorer, and then enter the IP address of the SPA9000. Click Admin
Login and then click Advanced.

STEP 2

Choose the client station that you want to configure:
a. Click the PBX Status link near the top right corner or lower left corner of the
page to view the list of client stations.
b. Click the hyperlink in the IP Address column for the first phone that you want to
configure. The telephone configuration page appears in a separate browser
window.

STEP 3

Configure the shared extension on the selected station:
a. Click the tab for the extension that you want to configure (Ext 1 ... Ext n,
depending on the phone model).

NOTE As a general practice, Cisco recommends that you always reserve

Extension 1 on the client station as the primary and private extension
of the designated user.
b. Scroll down to the Share Line Appearance section, and then enter the
following settings:

-

Share Ext: Choose shared.

-

Shared User ID: Enter a user ID number for this SLA. After the line key is
configured, this ID will appear on the phone display and in the Corporate
Directory.
SPA9xx Telephone Configuration > Ext: Shared Line Appearance

c. Scroll down to the Subscriber Information section, and then enter a User ID
and Display Name for this shared extension.

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SPA9xx Telephone Configuration > Ext: Subscriber Information

d. Scroll down the Proxy and Registration section, and enter the same Proxy
value as on the primary extension for this station.
SPA9xx Telephone Configuration > Ext: Proxy and Registration

STEP 4

Configure a line key button for the shared extension:
a. Click the Phone tab.
b. Scroll down to the Line Key area for the line key button that you want to
configure (Line 1 ... Line n, depending on the phone model).
c. Enter the following settings to correspond with the entries that you made on
the Extension tab:

-

Extension: From the drop-down list, choose the number corresponding
to the Ext N tab that you configured for this SLA. For example, if you
configured the SLA on the Ext 2 page, then choose 2 from the Extension
drop-down list.

-

Shared Call Appearance: From the drop-down list, choose shared.

-

Short Name: Type the extension number that you entered as the Shared
User ID for the extension. This extension number will appear on the
phone display and in the Corporate Directory.
SPA9xx Telephone Configuration: Ext: Line Key

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STEP 5

If needed, configure additional line keys for the same SLA.

STEP 6

Click Submit All Changes.

STEP 7

Repeat this procedure for each phone that you want to configure with the SLA.

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5
Administering the SPA400 and Voice Mail
Service

This chapter guides you through the process of configuring and managing the
SPA400 for PSTN access and voice mail service.
The SPA400 provides a SIP-PSTN gateway for voice connectivity between the
PSTN and the local client stations that are connected to the SPA9000. It also
includes an integrated voice mail application that supports up to 32 voice mail
accounts with customized greetings, providing the ability to receive and playback
voice mail messages.

•

“Connecting to the SPA400 Administration Web Server,” on page103

•

“Configuring the SPA400 Network Connection,” on page104

•

“Managing Access to the SPA400 Web-Based Configuration Utility,” on
page106

•

“Upgrading the Firmware for the SPA400,” on page107

•

“Configuring a SPA400 to Interoperate with the SPA9000,” on page108

•

“Configuring a SPA400 for Voice Mail Service,” on page112

•

“Managing the Voice Mail Messages on the USB Key,” on page121

•

“Enabling Debugging on the SPA400,” on page122

NOTE Also see:

•

Chapter 8, “Localization” for information about localizing your SPA400 devices

•

Appendix C, “SPA400 Field Reference” for detailed information about the fields
on each page of the SPA400 administration web server

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Connecting to the SPA400 Administration Web Server
You can manage a SPA400 by using the web-based configuration utility.

NOTE If you have not already done so, connect the SPA400 to the same switch as the

SPA9000 and the SPA900 series IP phones. For more information, see the SPA400
Quick Install Guide.

STEP 1

Start Internet Explorer, and enter the IP address of the SPA400 that you want to
configure.

NOTE By default, the SPA400 is configured to obtain an IP Address via DHCP. You

can check the obtained IP address on the router DHCP server’s client list.
STEP 2

When the password prompt appears, enter the default user name, Admin, with no
password. Then click OK.

NOTE The user name must be entered exactly as shown: Admin. By default, no

password is required, but a password can be set on the Administration >
Management page. See “Managing Access to the SPA400 Web-Based
Configuration Utility,” on page106.

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Configuring the SPA400 Network Connection
The SPA400 becomes a DHCP client of any server on the network. The
recommended setting is to use a static IP address. This configuration provides
ease of installation and prevents connectivity issues that would occur if the IP
address of the SPA400 changed.
STEP 1

Start Internet Explorer, and enter the IP address of the SPA400.

NOTE By default, the SPA400 is configured to obtain an IP Address via DHCP. You

can check the obtained IP address on the router DHCP server’s client list.
STEP 2

When the password prompt appears, enter the default user name, Admin, with no
password. Then click OK.

NOTE The user name must be entered exactly as shown: Admin. For information

about access, refer to “Managing Access to the SPA400 Web-Based
Configuration Utility,” on page106.
STEP 3

Click Setup tab > Basic Setup.

STEP 4

Enter the following settings:
Network Setup section:

•

Fixed IP address: Click the radio button, and then enter a valid IP address.

NOTE To avoid addressing conflicts, enter an IP address that is outside the

range of addresses that are automatically assigned by your DHCP
server.

•

IP Subnet Mask: Enter the subnet mask for the subnetwork that the
SPA400 is on.

•

Gateway IP Address: Enter the IP address of the router for this subnetwork.

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Saving or Discarding Changes on the SPA400

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Domain Name Server (DNS) Address section:

•

Primary DNS: Enter the IP address of the primary DNS server.

•

Secondary DNS: Enter the IP address of the secondary DNS server.

NTP section:

STEP 5

•

NTP: Enter a fully qualified name of a Network Time Protocol server, such as
time.nist.gov.

•

Time Zone: Select the time zone for your region.

Click Save Settings. The SPA400 will reboot. To reconnect to the web
administration server, enter the new IP address for the SPA400 in the browser
Address bar.

Saving or Discarding Changes on the SPA400
Changes can be saved or discarded at any time.

TIP

•

Changes are submitted only when you click the Save Settings button at the
bottom of a page.

•

To discard unsubmitted changes, click the Cancel Changes button at the
bottom of the page.

•

Unsubmitted changes are not retained when you move among the pages.

•

After you submit changes in the SIP or Voice mail accounts settings, reboot the
SPA400.

Before you make changes, save a copy of your current working configuration:
1. In Internet Explorer, connect to the administration web server.
2. Navigate to the page that has the settings that you want to save.
3. From the menu, choose File > Save As.

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4. Save the configuration as Web Page Complete. The currently displayed page is
saved. You can use the saved file to review the settings as needed.
5. Repeat these steps for each page, as needed.

Managing Access to the SPA400 Web-Based Configuration
Utility
One log on can be established for access to the SPA400 web-based configuration
utility. The default username of Admin can be changed, and a password can be
entered.
STEP 1

Connect to the SPA400 administration web server. (See “Connecting to the
SPA400 Administration Web Server,” on page103.)

STEP 2

Click Administration tab > Management.
SPA400 Administration > Management

STEP 3

Proceed as needed:

•

Gateway Username: Type the desired username in this field, up to 32
characters.

•

Gateway Password: Type the password.

•

Retype to Confirm: Type the password again. Both entries must match exactly.

STEP 4

Click Save Settings.

STEP 5

To restart the SPA400, complete the following steps:
a. Click Administration tab > Reboot.
b. Click the Restart System button.
c. When the confirmation message appears, click OK. The SPA400 reboots.

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5

d. When the Reboot OK . Go to Setup page? message appears, click OK and wait
for 60 seconds (the time required for the SPA9000 to re-register with SPA400).

Upgrading the Firmware for the SPA400
As needed, you can download new firmware and then install it on the SPA400.
STEP 1

Download the latest SPA400 firmware from the following URL:
http://tools.cisco.com/support/downloads/go/Redirect.x?mdfid=282414117

STEP 2

Connect to the SPA400 administration web server. (See “Connecting to the
SPA400 Administration Web Server,” on page103.)

STEP 3

Click Administration tab > Firmware Upgrade.

STEP 4

Click Browse.
SPA400 Administration tab > Firmware Upgrade page

STEP 5

Find the binary (.bin file) that you extracted to your Desktop, and click Open. The
selected file appears in the File Path field on the Firmware Upgrade page.

STEP 6

Click Upgrade.

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STEP 7

When the confirmation message appears, click OK.

STEP 8

When the Setup page reappears, verify that the Firmware Version number
matches the firmware version that you installed. You have successfully upgraded
the firmware.
SPA400 Main Page: Firmware Version

NOTE You may need to refresh your browser to display the updated banner

reflecting the new version number.

Configuring a SPA400 to Interoperate with the SPA9000
A SPA400 must be configured to register the SPA9000. To enable the
interoperation of the two devices, you need to enter corresponding information on
the SPA9000 Voice > Line page and on the SPA400 Setup > SPA9000 Interface
page. For voice mail service, additional entries are needed on the SPA400 Setup
> Voicemail Server page.
Be aware of the following factors:

•

You need to complete this procedure for each SPA400, whether it is used as a
PSTN gateway or as a voice mail server.

•

You need to configure a SPA9000 line interface for each SPA400. See
“Configuring a Line Interface for a SPA400 (PSTN or Voice Mail),” on page 60.

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TIP

5

If you install multiple SPA400 units, keep track of the MAC addresses to ensure that
you know which device you are configuring. In the administration web server, you
can see the MAC address by clicking the Status tab.

STEP 1

Connect to the SPA400 administration web server. (See “Connecting to the
SPA400 Administration Web Server,” on page103.)

STEP 2

When the password prompt appears, enter the user name and password. Then
click OK.

NOTE For more information about the administrator account, see “Managing

Access to the SPA400 Web-Based Configuration Utility,” on page106.
STEP 3

Click Setup tab > SPA9000 Interface.

STEP 4

Enter the following settings:

•

User ID: 9000
This is the user ID that the SPA9000 will use to register with the SPA400. Any ID
can be used, but must match exactly the User ID that you entered on the
corresponding SPA9000 Voice > Line page.

•

SPA9000 Address: Select the Discover Automatically radio button (required
for SPA9000 operation).
This setting enables the SPA400 to learn the IP address and the UDP port of
the SPA9000 the from the SIP Registration packets sent by SPA9000.

•

Call Signalling Packets: 68

•

RTP Packets: b8

•

Leave the Signalling Port at the default value of 5060. This port is the source
port that is used to originate signaling between the SPA400 and the SPA9000.

•

Leave the RTP Port at the default value of 10000. This port is the base UDP port
for the block of UDP ports that the SPA400 uses to send and receive RTP and RTCP
packets.

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•

Leave the Session Timer fields at the default values: Enabled with 0 seconds in
both Refresh Time fields.

STEP 5

Click Save Settings at the bottom of the page.

STEP 6

Click Setup tab > Voice.

STEP 7

Enter the following settings:

•

Preferred Codec: Select G.711u.

•

Packetization: Choose 30ms from the Packetization drop-down list for G.711U.

•

VAD: Choose OFF from the VAD drop-down list for G.711U.
SPA400 Setup > Voice

STEP 8

Click Save Settings at the bottom of the page.

STEP 9

To restart the SPA400, complete the following steps:
a. Click Administration tab > Reboot.
b. Click the Restart System button.
c. When the confirmation message appears, click OK. The SPA400 reboots.
d. When the Reboot OK . Go to Setup page? message appears, click OK.

STEP 10 If you are using this SPA400 as a PSTN gateway, connect the RJ11 cables from the

SPA400 to the wall outlet for PSTN access.
STEP 11 To verify your progress, perform the following tasks:

•

Click the Status tab, and confirm that the SIP registration status is Registered.

•

If you connected PSTN lines to the ports on the SPA400, confirm that you can
place an external call to the phone number that is associated with the PSTN

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5

line. The call is directed according to the Contact List. Also, in the SPA400 web
configuration page, click the Status tab, and then verify that a voltage value
appears in the Battery Level section.
SPA400 Administration > Status: Battery Level

NOTE If the battery level is 0 V on a line that you have connected to the PSTN,

troubleshoot the phone wiring.

•

Connect to the SPA9000 administration web server, and choose Admin access
with Advanced settings. Click Voice tab > Info. Scroll down to the Line 2
Status section, and verify that the Registration Status is Registered.

STEP 12 Click Save Settings.

NOTE If you are using this SPA400 for voice mail service, continue to “Configuring

a SPA400 for Voice Mail Service,” on page112.

Configuring a SPA400 for PSTN Access
Typically, there is no need to change the default settings on the Voice page. Make
changes only if a problem is suspected and only after consulting with a service
technician from your telephone service provider. It is essential that the settings on
your system are compatible with those of the Central Office.

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5

Configuring a SPA400 for Voice Mail Service
The SPA400 includes a USB adapter with an integrated voice mail application for
the users and extensions that are configured on the SPA9000. The integrated
voice mail application server supports 32 configurable voice mail accounts.
Although a SPA9000 can be configured with up to four SPA400 devices, only one
SPA400 can be configured with the voice mail server.

NOTE Important: For optimum voice mail performance, a SPA400 should be dedicated to

the voice mail application when either of the following conditions is met:
1) More than 2 FXO connections are required
—OR—
2) More than 2 users commonly access voice mail at the same time.
This section includes the following topics:

•

“Voice Mail Capacity,” on page112

•

“Configuring Local Voice Mail Service on a SPA400,” on page113

•

“Setting Up Voice Mail on Each Station,” on page116

•

“Enabling Remote Voice Mail Access (Optional),” on page119

NOTE Before you begin any of the procedures in this section, configure a SPA9000 line

interface for this device. See “Configuring a Line Interface for a SPA400 (PSTN or
Voice Mail),” on page 60.

Voice Mail Capacity
The provided 128-Mb USB drive can store more than 3.8 hours of messages
(approximately 230 60-second messages). For example, with 10 voice mail users,
each user can store up to 23 60-second messages. When disk capacity is
reached, inbound voice mail deposit attempts are disconnected. If additional voice
mail storage is required, please contact Linksys for recommended high-capacity
USB drives.

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Configuring a SPA400 for Voice Mail Service

Configuring Local Voice Mail Service on a SPA400
You need to configure the voice mail server and set up the voice mail boxes for the
users.

NOTE The settings on this page correspond to the settings on the SPA9000 Voice > Line

page. See “Configuring a Line Interface for a SPA400 (PSTN or Voice Mail),” on
page 60.

SPA400 > Setup > Voicemail Server
SPA9000 > Voice > Line N

STEP 1

Connect to the SPA400 administration web server. (See “Connecting to the
SPA400 Administration Web Server,” on page103.)

STEP 2

Click Setup tab > Voicemail Server.

STEP 3

Enter the following information:

•

Server Port: 5090
The voice mail server uses this UDP port to listen for signalling between the
SPA400 and the SPA9000. This port must be different from the port number
that you entered on the SPA9000 Interface page.

•

SPA9000 subscriber ID: 8888
The SPA9000 uses the subscriber ID to subscribe to the SPA400 Voice mail
Server for obtaining notification.

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•

Mailbox deposit number: 900
The SPA9000 uses the deposit number to deposit voice mail on the voice mail
server.

•

Mailbox manage number: 800
The SPA9000 uses the deposit number to access voice mail on the voice mail
server.
SPA400 Setup > Voicemail Server: Voicemail Setting

STEP 4

Click Save Settings.

STEP 5

Configure the voice mail users (required):
a. Click the Voicemail Users tab.
b. Enable the voice mail accounts and enter the user’s extensions and passwords:

-

Enable: Select the check box to enable the voice mail account.

-

User ID: Enter the user’s extension number.

-

Password: Enter a password for this user. Users can change their own
passwords after logging on with the assigned password.
SPA400 Setup > Voicemail Users

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NOTE Later you configure each client station for voice mail access to the

mail boxes that you create on this page.
STEP 6

Click Save Settings at the bottom of the page.

STEP 7

Restart the SPA400 by completing the following steps:
a. Click Administration tab > Reboot.
b. Click the Restart System button.
c. When the confirmation message appears, click OK. The SPA400 reboots.
d. When the Reboot OK . Go to Setup page? message appears, click OK.

STEP 8

To verify your progress, click the Status tab, and verify the following settings:

•

USB status: Mount

•

Voice mail status: OK

•

SPA9000 Registration status: Registered
SPA400 Status

NOTE If the SPA registration status is not Registered, trying powering off the

SPA9000 and powering it on again.

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Configuring a SPA400 for Voice Mail Service

STEP 9

Continue to the next procedure, “Setting Up Voice Mail on Each Station,” on
page116.

Setting Up Voice Mail on Each Station
You need to set up each station that needs to have a voice mailbox. Perform this
procedure after you have configured both the SPA400 and the SPA9000.
STEP 1

Connect to the SPA9000 administration web server. (See “Connecting to the
SPA9000 Administration Web Server,” on page 27.)

STEP 2

Click the PBX Status link near the top right corner or lower left corner of the page.
The screen lists each phone by Station Name and Extension number.
SPA9000 PBX Status

STEP 3

Find the phone that you want to configure, and then click the hyperlink in the IP
Address column. The Telephone Configuration page appears in a separate
browser window.
SPA9xx Telephone Configuration > Info

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STEP 4

5

To assign a station name, complete the following steps:
a. Click the Phone tab.
b. In the General section, type a name in the Station Name field.
SPA9xx Telephone Configuration > Phone: General

NOTE This setting assists you in managing the phones. The station name appears

on the phone display, in the Corporate Directory, and in features such as
Group Call Pickup, that list the participating phones in a menu.
STEP 5

To assign a voice mail box to the primary extension, complete the following steps:
a. Click the Ext 1 tab.
b. Scroll down to Call Feature Settings.
c. Enter the Mailbox ID in the following format:


Example: 2105

-

lineN: The SPA9000 line (1, 2, 3, or 4) that is configured with the voice
mail settings
In the example, 2 is the number of the SPA9000 line interface that is
configured for the SPA400 voice mail server.

-

mailbox: The voice mailbox number for this station, as configured on the
SPA400 Voice Mail Users page.
In the example, voice mailbox 105 is assigned to the station.

NOTE The mailbox value corresponds to a User ID on the SPA400 voice mail

server.

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SPA400 > Setup > Voicemail Users

193968

Phone Configuration > Ext N

STEP 6

Click Submit All Changes. The phone reboots.

STEP 7

Close the browser window for this station.

STEP 8

Return to the browser window that shows the list of stations, and then repeat this
procedure for each station that you need to configure.

NOTE When you finish configuring stations, you can click the Back button on the

browser toolbar to return to the main web configuration page.
STEP 9

To verify your progress, perform the following tasks:

•

Station Name: Verify that the station name appears on the phone display and
in the list of stations on the PBX Status page.

•

Mailbox Status: Connect to the SPA9000 administration web server, and
choose Admin access with Advanced settings. Click Voice tab > Line 2. In the
Proxy and Registration section, check the Mailbox Status field. You should see
a listing for each extension that you configured with voice mail. Refer to the
following example.

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SPA9000 Voice > Line: Proxy and Registration

•

Voice Mail: Press the Message button. You hear one of the following
responses:

•

“Password”: If you are prompted for a password, the station is configured
properly. You can enter the password and manage the mailbox.

•

“Mailbox number”: If you are prompted for the mailbox number, a mailbox is
not assigned to this station. Review the settings on the SPA400 Voice mail
User page (see “Configuring Local Voice Mail Service on a SPA400,” on
page113) and the Phone configuration page (see “Setting Up Voice Mail on
Each Station,” on page116).

•

Busy Tone: A busy tone indicates a problem with the configuration. Verify
that the USB drive is properly inserted into the SPA400, and review the
various entries that you made in the configuration.

Enabling Remote Voice Mail Access (Optional)
You can configure the SPA9000 to allow the users to check their voice mail when
they are out of the office.

NOTE If your users will call into your voice mail system through an ITSP line, your ITSP

must support out-of-band DTMF (IE RFC2833).

STEP 1

Connect to the SPA9000 administration web server. (See “Connecting to the
SPA9000 Administration Web Server,” on page 27.)

STEP 2

Click Voice tab > SIP.

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STEP 3

Scroll down to the Auto Attendant Parameters section.

STEP 4

Edit the AA Dial Plan 1 string to include a code for the voice mail server, as
described below.

5

•

SYNTAX: (10x|xxx.|)

•

EXAMPLE: (10x|xxx.|<8:vmm2>)

•

dialcode: The digit that users dial, when prompted by the Auto Attendant, to
access voice mail remotely.

NOTE The Contact List for this line must be configured for the Auto Attendant to

answer.

•

MailboxManageNumber: The Mailbox Manage Number that was entered on the
SPA400 Voice mail Settings page. If you followed the instructions in the
procedure “Configuring a SPA400 to Interoperate with the SPA9000,” on
page108, you set 800 as the Mailbox Manage Number.

•

vmmN: Replace N with the number of the SPA9000 line interface (Line 1 ... Line
4) that is configured for the SPA400 voice mail server.

STEP 5

Copy and paste the same string into the AA Dial Plan 2 field, for the purpose of
allowing remote access of voice mail at all times of day. For more information
about AA Dial Plans, refer to “Configuring Dial Plans for the Auto Attendant,” on
page150.

STEP 6

Click Voice tab > Line N, where N is the number of the line interface for this
SPA400 unit.

STEP 7

Make sure that the VMSP Bridge, XFER Bridge Mode, and CFWD Bridge Mode
fields are set to all.

STEP 8

Click Submit All Changes. The SPA9000 and the phones reboot.

STEP 9

To verify your progress, perform the following tasks:
a. Dial into the site from an external number.
b. When the Auto Attendant prompts you for an extension, press 8.
c. When the Voice Mail Server prompts for a mailbox number, enter the mailbox
number.
d. When prompted for a password, enter the password.

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Managing the Voice Mail Messages on the USB Key

5

Managing the Voice Mail Messages on the USB Key
To delete unneeded voice mail messages on the USB key, refer to the procedure
below.

NOTE When you click the Reset button, all the voice mail message for all users will be

deleted. The deleted voicemail messages are not recoverable.

STEP 1

Connect to the SPA400 administration web server. (See “Connecting to the
SPA400 Administration Web Server,” on page103.)

STEP 2

Click Administration tab > USB Setting.
SPA400 Administration > USB Setting

STEP 3

Check the I want to reset USB check box.

STEP 4

Click Reset.

STEP 5

When the confirmation message appears, click OK to continue or click Cancel to
end the operation without deleting the messages.

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Enabling Debugging on the SPA400

5

Enabling Debugging on the SPA400
If you are investigating issues, you can collect system logs and debug information
for the SPA400.
Requirements:

•

You need a PC that is on the same subnetwork as the SPA9000, to capture the
log files. This PC needs to be running a syslog daemon. Enter the IP address of
this PC on the Voice > System page, in the Syslog Server and Debug Server
fields.

•

You can deploy a syslog server to receive syslog messages from the device,
which acts as a syslog client. The syslog client device uses the syslog protocol
to send messages, based on its configuration, to a syslog server. The syslog
messages can be accessed by reviewing the "syslog.514.log" file which
resides in the same directory as the slogsrv.exe syslog server application.

NOTE Partners can download the Syslog Server for SPA Devices by going to Cisco

Partner Central, Voice & Conferencing page, Technical Resources section.
Use the following URL:
www.cisco.com/web/partners/sell/smb/products/
voice_and_conferencing.html#~vc_technical_resources

STEP 1

Connect to the SPA400 administration web server. (See “Connecting to the
SPA400 Administration Web Server,” on page103.)

STEP 2

Click Event Logs tab.

STEP 3

Use the drop-down lists to choose the types of information that you want to
collect.

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Enabling Debugging on the SPA400

SPA400 Event Logs

STEP 4

Click Save Settings.

NOTE As a best practice, enable logging only when needed, and disable logging

when you finish the investigation. Logging information can impact system
performance.
STEP 5

Click Setup tab > Basic Setup.

STEP 6

In the Syslog Settings section, enter the IP address of the Syslog Server, which
normally is a PC on the same network as the SPA400.

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Enabling Debugging on the SPA400

SPA400 Setup > Basic Setup

STEP 7

Start a command prompt on the PC, and run the system logging software.

STEP 8

Run the scenario that causes the problem and when done, press CTRL+C in the
command window to stop the logs.

STEP 9

When you are finished collecting logs, return to the Event Logs page and reset the
logs to Off.

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Configuring Music on Hold

This chapter explains how to configure Music on Hold using either a music file or
streaming audio.
This chapter includes the following topics:

•

“Using the Internal Music Source for Music On Hold,” on page125

•

“Configuring a Streaming Audio Server,” on page127

Using the Internal Music Source for Music On Hold
An internal music source with the user ID imusic is available. It plays an internally
stored music file repeatedly. The unit ships with a default music file (Romance de
Amor). You can override this file by downloading a new file into the unit by using
TFTP.
Refer to the following topics:

•

“Using the Internal Music Source,” on page125

•

“Changing the Music File for the Internal Music Source,” on page126

Using the Internal Music Source
To use the internal music source, simply identify imusic as the MOH server for each
IP phone.
STEP 1

Use the phone menu to find the IP address of the phone:
a. Press the Setup button on the phone keypad.
b. Press 9 - Network, and then scroll down to 2- Current IP Address.

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6

STEP 2

Start Internet Explorer, and then enter the IP address of the telephone. The
Telephone Configuration page appears in a separate browser window.

STEP 3

Click Admin Login, and then click Advanced.

STEP 4

Click the Ext 1 tab.

STEP 5

Scroll down to the Call Feature Settings section.

STEP 6

Enter the following value in the MOH Server field: imusic

STEP 7

Click Submit All Changes.

STEP 8

To verify, place a test call to the extension. When the call is answered and put on
hold, the caller should hear the default music file (Romance de Amor).

Changing the Music File for the Internal Music Source
The following resources are required to change the music file for the internal music
source:

•

TFTP server software

•

The IP address of the administration computer that is connected to the
SPA9000

•

A music source in G.711u format, sampled at 8000 samples/sec, up to 65.5
seconds in length, with no header information

STEP 1

Before you begin, make sure that you have TFTP server software running on your
computer.

STEP 2

Start Internet Explorer, connect to the administration web server, and choose
Admin access with Advanced settings.

STEP 3

Click Voice tab > SIP.

STEP 4

Scroll down to the Internal Music Source Parameters section.

STEP 5

Enter the following URL in the Internal Music URL field:
tftp://server_IPaddress:portpath

•

server_IPaddress: The local IP address of the computer you are using as the
TFTP server

•

port: The port number used by the TFTP server (default 69)

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Configuring a Streaming Audio Server

STEP 6

6

•

path: The location and name of a music file in the correct format

•

For example, if the computer local IP address is 192.168.0.5, the directory is
named musicdir, and the converted music file is named jazzmusic.dat , then you
would enter the following URL: tftp://192.168.0.5:69/musicdir/
jazzmusic.dat

Click Submit All Changes. The unit reboots. Then the unit downloads the file and
stores it in flash memory.

Configuring a Streaming Audio Server
This section describes how to use and configure a streaming audio server (SAS). It
includes the following topics:

•

“About the Streaming Audio Server,” on page127

•

“Configuring the Streaming Audio Server,” on page129

•

“Using the IVR with an SAS Line,” on page130

About the Streaming Audio Server
The Streaming Audio Server (SAS) feature lets you attach an audio source to an
FXS port and use it as a streaming audio source device. If the unit has multiple FXS
ports, either or both of the associated lines can be configured as an SAS server.
Use a media signal adapter or “music coupler” to connect an Ethernet cable from a
media source to the FXS port. For example, the MC-9700 Music Coupler has been
tested with ATA devices and is available at the following URL:
www.neogadgets.com/cart/
cart.php?target=product&product_id=17&substring=music+coupler

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Configuring Music on Hold
Configuring a Streaming Audio Server

After you complete the required configuration, the FXS port is ready to stream
audio. The functionality depends on the hook state of the FXS port:

•

If the FXS port is off hook, an incoming call is answered automatically and
audio is streamed to the calling party.

NOTE Each SAS server can maintain up to five simultaneous calls. If the

second line on the unit is disabled, then the SAS line can maintain up
to 10 simultaneous calls. Further incoming calls receive a busy signal
(SIP 486 Response).

•

If the FXS port is on-hook when the incoming call arrives, a SIP 503 response
code is transmitted to indicate “Service Not Available.”

•

If an incoming call is auto-answered, but later the FXS port changes to on-hook,
the call is not terminated but continues to stream silence packets to the caller.

•

The SAS line can be set up to refresh each streaming audio session
periodically using a SIP re-INVITE message, which detects if the connection to
the caller is down. If the caller does not respond to the refresh message, the
SAS line terminates the call so that the streaming resource can be used for
other callers.

Additional information:

•

The SAS line does not ring for incoming calls even if the attached equipment is
on-hook.

•

If no calls are in session, battery is removed from tip-and-ring of the FXS port.
Some audio source devices have an LED to indicate the battery status. This can
be used as a visual indication as to whether audio streaming is in progress.

•

Call Forwarding, Call Screening, Call Blocking, DND, and Caller-ID Delivery
features are not available on an SAS line.

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Configuring the Streaming Audio Server
Use the following procedure to configure an SAS with an external music source.
STEP 1

Connect an RJ-11 adapter between the music source (a CD player or iPod, for
example) and an FXS port.

STEP 2

Start Internet Explorer, connect to the administration web server, and choose
Admin access with Advanced settings.

STEP 3

Configure the FXS port:
a. Click Voice tab > FXS N, where N represents the number of the FXS port
where you connected the cable from the external music source.
b. In the Subscriber Infomation section, enter the following settings:

•

Display Name: Enter an extension number of name for the FXS 1 port, such
as Receptionist Area Fax Machine.

•

User ID: Enter a three- to four-digit extension number that is not is use by
another extension.

c. In the Streaming Audio Server (SAS) section, choose yes from the SAS
Enable drop-down list.
STEP 4

Click Submit All Changes.

STEP 5

Configure each phone to use this audio source as the MOH server:
a. Click the PBX Status link to view the list of phones.
b. In the list, find the phone that you want to configure, and then click the hyperlink
in the IP Address column. The Telephone Configuration page appears in a
separate window.
c. Click the Ext 1 tab.
d. Scroll down to the Call Feature Settings section.
e. In the MOH Server field, enter the extension number that you assigned to the
FXS port for the streaming audio server.
f.

Click Submit All Changes.

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g. Close the window for the Telephone Configuration page.
h. Repeat this step to configure each phone, as needed.

Using the IVR with an SAS Line
The IVR can still be used on an SAS line, but the user needs to follow the following
steps:
STEP 1

Power off the SPA9000.

STEP 2

Connect a phone to the port and make sure the phone is on-hook.

STEP 3

Power on the SPA9000.

STEP 4

Pick up handset and press * * * * to invoke IVR in the usual way.
If the SPA9000 boots and finds that the SAS line is on-hook, it does not remove
battery from the line so that IVR may be used. But if the SPA9000 boots up and
finds that the SAS line is off-hook, it removes battery from the line because no
audio session is in progress.

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Configuring the Auto Attendant

This chapter describes how to configure the SPA9000 Voice System Auto
Attendant (AA) by using the IVR and XML scripting.
This chapter contains the following sections:

•

“How the Auto Attendant Works,” on page131

•

“Working with the Auto Attendant Greetings,” on page133

•

“Writing an Auto Attendant Script,” on page138

•

“Configuring the DayTime, NightTime and Weekend/Holiday Auto Attendants,”
on page148

•

“Configuring Dial Plans for the Auto Attendant,” on page150

How the Auto Attendant Works
The Auto Attendant (aa) is an internal service within the SPA9000. This service
answers calls and plays pre-recorded voice messages that offer a menu of
choices. The Auto Attendant parses the user input (key presses or DTMF tones),
based on the Auto Attendant Dial Plan. Finally, the Auto Attendant routes the call to
the selected extension.
The Auto Attendant can handle up to ten incoming calls simultaneously. It can
accommodate two categories of callers:

•

Callers who do not call you regularly or are not accustomed to using AutoAttendants. The Auto Attendant plays a greeting and provides menus to help
callers reach the desired extension.

•

Callers who call you regularly and want to move through the system quickly.
The Auto Attendant allows experienced users to input their responses at any
time.

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How the Auto Attendant Works

7

The Auto Attendant relies upon the following components:

•

Contact List. When a call comes in, the SPA9000 directs the call according to
the entries in the Contact List for the line interface. By default, the Auto
Attendant is the only client on this list, so the Auto Attendant picks up every
call. Alternatively, you can enter a list of client stations to alert, with the Auto
Attendant picking up a call only if a number (or group) of clients did not pick up
the call first. This parameter is configured on the Voice > Line N page,
Subscriber Information section, Contact List field. For more information, see
“Managing Inbound Calls with the Contact List,” on page 85.

•

Recorded Prompts. The Auto Attendant plays prompts to assist the users. The
system includes a set of pre-recorded prompts, and you can record your own
custom prompts. For more information, see “Working with the Auto Attendant
Greetings,” on page133.

•

AA Script. This XML script determines which prompt is used to greet callers
and which prompts are played in response to valid or invalid user inputs. You
can enter up to three scripts. One script is active at any time. These scripts are
configured on the Voice > SIP page, Auto Attendant Parameters section, AA
Script 1 - 3 fields. For more information, see “Writing an Auto Attendant Script,”
on page138.

•

DayTime, NightTime, and Weekend/Holiday AA. You can activate different
AA scripts for different times of day. By default, the DayTime AA settings are
activated and use AA Script 1. As needed, you can activate the NightTime AA
and the Weekend/Holiday AA, with your selection of a script (AA Script 1, 2, or
3). You also need to define the start and end time for daytime, nighttime, and
weekend/holiday. For more information, see “Configuring the DayTime,
NightTime and Weekend/Holiday Auto Attendants,” on page148.

•

AA Dial Plan. The Auto Attendant parses the user input according to the dial
plan that is identified in the AA script. You can create up to two dial plans.
These dial plans are configured on the Voice > SIP page, Auto Attendant
Parameters section, AA Dial Plan 1 - 2 fields. See “Configuring Dial Plans for
the Auto Attendant,” on page150.

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Configuring the Auto Attendant
Working with the Auto Attendant Greetings

Working with the Auto Attendant Greetings
This section provided information about adding, and editing the Auto Attendant
greetings. See the following topics:

•

“Using Pre-Recorded Prompts,” on page133

•

“Recording an Auto Attendant Prompt,” on page134

•

“Downloading Prompts,” on page137

Using Pre-Recorded Prompts
You can save up to ten Auto Attendant prompts. Four pre-recorded prompts are
provided, as listed in the following table.
Table 1 Default AA Prompts

Prompt ID

Default Audio Content

Prompt1

“If you know your party’s extension, you may enter it now.”

Prompt2

“Your call has been forwarded.”

Prompt3

“Not a valid extension, please try again.”

Prompt4

“Goodbye.”

NOTE These prompts are available in languages other than English. See “Localizing the

SPA9000 Auto Attendant Prompts,” on page151.
You can replace the pre-recorded prompts with your own recordings, and you can
add up to six additional recordings. For example, you may want to change the
default prompt, Prompt 1, to greet callers with your company name. You may want
to record a different prompt for nighttime or holiday hours. You can make these
recordings by using the IVR. The recordings are encoded with G.711U and saved
in flash.

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Working with the Auto Attendant Greetings

7

Recording an Auto Attendant Prompt
Follow this procedure to overwrite an existing recording or to add a new
recording.

NOTE Customized prompts are erased when a factory reset is performed on the

SPA9000.

STEP 1

Connect an analog phone to the Phone 1 or Phone 2 port of the SPA9000.

STEP 2

Press the star key (*) four times: ****
The IVR plays the following prompt: Linksys configuration menu. Please enter the
option followed by the # (pound) key or hang up to exit.

STEP 3

Press 72255# to access the Auto Attendant message settings.
The IVR plays the following prompt: Please enter the message number followed
by the # (pound) key.

STEP 4

Enter the number of the message (1 through 10) that you wish to record, review, or
delete.
The IVR plays the following prompt: Enter 1 to record. Enter 2 to review. Enter 3
to delete. Enter * to exit.

STEP 5

Press 1 to record a new message.

NOTE The IVR checks the available buffer size. If there is no more buffer capability,

IVR plays the Option Failed message and returns to the previous menu.
If the buffer space is sufficient, the IVR plays the following prompt: You may
record your message after the tone. When finished, press #.
STEP 6

After the tone, record the new message and then press #.
The IVR plays the following prompt: To save, enter 1. To review, enter 2. To rerecord, enter 3. To exit, enter *.

STEP 7

Press 1 to save the new recorded message.
The IVR plays the following prompt: One moment, please.

STEP 8

Wait for several seconds while the save is completed.

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Working with the Auto Attendant Greetings

Example Prompts
In this example, the business will have different Auto Attendant Scripts for different
times of day.
Table 2

Example AA Prompts

Prompt ID

Message

Prompt1

“If you know your party’s extension, you may enter it now.”

Prompt2

“Your call has been forwarded.”

Prompt3

“Not a valid extension, please try again.”

Prompt4

“Goodbye.”

Prompt5

“Welcome to All Seasons Travel.”

Prompt6

“Thank you for calling All Seasons Travel. Presently we are
closed.”

Prompt7

“We are open Monday through Friday 9 AM to 6 PM, and we
are closed on Saturdays and Sundays. Our address is 101
Main Street, Anytown, Anystate, USA.”

Prompt8

“If you know your party’s three-digit extension, you may
enter it now.”

Prompt9

“To reach our receptionist, press 0 at any time. For our
company location, press 1. For travel support, press 2. For
sales, press 3. Otherwise, please stay on the line for our
receptionist.”

Prompt10

“Welcome to the All Seasons Travel support line. If this is
regarding our Holiday Getaway Special, please press 1;
otherwise, please stay on the line for one of our travel
associates. Or press * to go back.”

For examples of AA scripts that use custom prompts, see “An Introduction to XML
Scripting Grammar in AA Script Examples,” on page138.

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Working with the Auto Attendant Greetings

Using the IVR Prompts to Change Recordings
The IVR prompts guides you through the process of recording, erasing, and
changing greetings. Refer to the following illustration.
Figure 1

IVR Prompt Menu Call Flow
User enters
72255 from IVR
Menu-1
Play prompt (“Please enter message
number followed by the # key”)

User enters
prompt key
(1-10)
Menu-2
Play prompt (“Enter 1 to record,
enter 2 to review, enter 3 to
delete, enter * to exit”)

Goto menu-2

Goto menu-1

User enters *
User enters 2

User enters 1

Retrieve the
prompt data and
play prompt

User enters 3

Menu-3-A
Play prompt (“You may record your
message after the tone. When
finished, press #.”)

User enters 2

Play prompt
(“Value saved”)

SPA9000 Voice System Administration Guide

Play user
recorded
prompt

to exit

User enters *

Erase message

Menu-4-A
Play prompt (“to save enter 1, to
review enter 2, to rerecord enter
3, to exit enter *.”)

User enters 1

confirm, enter *
User enters 1

User records
a prompt

Goto
menu
4-A

Menu-3-B
Play prompt (“enter 1 to

Play prompt
(“Message erased.”)

User enters 3

Goto menu-3-A

User enters *

Goto menu-2

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Configuring the Auto Attendant
Working with the Auto Attendant Greetings

7

Downloading Prompts
You can download customized prompt files from a TFTP/HTTP/HTTPS server.
These files must be encoded in G.711u, size less than 60 seconds, with the header
removed. The total prompt file size cannot be larger than 94.5 seconds.
The prompt is downloaded at the device boot up time. If the prompt has already
been downloaded from the given URL, the download is not performed. If the
prompt file name is none, the corresponding prompt currently saved in the flash is
erased. The default value is blank.
STEP 1

Connect to the administration web server, and choose Admin access with
Advanced settings.

STEP 2

Click Voice tab > SIP.

STEP 3

Scroll down to the Auto Attendant Parameters section.

STEP 4

In the AA Prompts URL Script field, enter the location and the file name for the
prompts files, in the following format:
SYNTAX:
serv=scheme://server_addr[:port]/root_path;[p1={prompt1 file
path name};][p2={prompt2 file path name};][p3={prompt3 file
path name};][p4={prompt4 file path name};][p5={prompt5 file
path name};][p6={prompt6 file path name};][p7={prompt7 file
path name};][p8={prompt8 file path name};][p9={prompt9 file
path name};][p10={prompt10 file path name};]

•

scheme = tftp|http|https

•

default port is 69 for tftp, 80 for http, and 443 for https

•

root_path can be empty

•

[ ] denotes optional item

•

none: The prompt will be erased.

EXAMPLE:
serv=tftp://192.168.2.150/root/
test;p1=menu.wav;p2=transfer.wav; p3=nomatch.wav;p4=none;
In this example, Prompt 1 is downloaded from tftp://192.168.2.150/root/test/
menu.wav, Prompt 2 from tftp://192.168.2.150/root/test/transfer.wav, and Prompt 3
from tftp://192.168.2.150/root/test/nomatch.wav. Prompt 4 is erased.

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Configuring the Auto Attendant
Writing an Auto Attendant Script

STEP 5

Click Submit All Changes.

Writing an Auto Attendant Script
The SPA9000 AA allows users to define the AA instructions using XML script. This
section includes the following topics:

•

“An Introduction to XML Scripting Grammar in AA Script Examples,” on
page138

•

“Elements of XML Scripting Grammar ,” on page142

•

“Auto Attendant XML Instructions Set,” on page145

•

“Entering an Auto Attendant Script,” on page147

An Introduction to XML Scripting Grammar in AA Script
Examples
The SPA9000 lets you use XML scripting grammar to define the Auto Attendant
instructions. You can study the example scripts to learn about the scripting
grammar. You also may find it helpful to use these scripts as the basis for your own
custom scripts. Also see “Elements of XML Scripting Grammar ,” on page142.

Example 1: Routing Calls to Any Extension Number (Default AA Script 1)
In this example, the business is using the default AA Script 1, which prompts the
caller to enter any extension number. Each part of the script is described in the
following table.
Table 3

Elements of the Default AA Script 1

Script Elements

Purpose



This script is for the AA.



This form is given the name “dir” for
directory, and it is a menu type of form,

which accepts DTMF inputs.

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Configuring the Auto Attendant
Writing an Auto Attendant Script

Script Elements

Purpose



SPA9000 Voice System Administration Guide

Not a valid extension, please try
again.

The form is ended.
The script is ended.

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Configuring the Auto Attendant
Writing an Auto Attendant Script

Example 2: Routing Calls with a Departmental Sub-Menu
In this example, the business wants to expedite the handling of sales calls. A
custom prompt is used to give special instructions about calls for the sales
department. A sub-menu for the sales group is provided to direct calls to the
correct person within that department.
Table 4 Elements of an AA Script with a Sub-Menu

Script Elements

Purpose



This script is for the AA.



”

•

If the user presses *, the Auto
Attendant plays Prompt 4: Goodbye.

•

If the user presses any digits other
than *, the Auto Attendant plays
Prompt 2 ( Your call has been
transferred.) The call is transferred
to the extension number that the user
entered.

The form is ended.
The script is ended.

Elements of XML Scripting Grammar
This section includes the following topics:

•

“Audio Instruction,” on page142

•

“Action Instruction,” on page143

•

“Noinput Instruction ,” on page143

•

“Nomatch Instruction,” on page144

•

“Menu Matched Instruction for Touch Tone (DMTP) Input,” on page144

Audio Instruction
The following is an example of the audio instruction:

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