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ADMINISTRATION
GUIDE

Linksys Voice System
SPA9000 IP Telephony System
SPA400 PSTN VoIP Gateway with Integrated Voice Mail Server
SPA9x2 IP Phones

Table of Contents

Preface . . . . . . . . . . . . . . . . . . . . . . . . . . . . . vi
Document Audience
Firmware
Document Purpose and Contents
Document Conventions
Related Documentation
Finding Information in PDF Files
Finding Text in a PDF
Finding Text in Multiple PDF Files
Online Resources
Copyright and Trademarks
Technical Support

vi
vii
viii
ix
ix
xi
xi
xi
xiii
xiii
xiii

Chapter 1: Getting Started . . . . . . . . . . . . . . . . . . . 13
Introduction to the Linksys Voice System
SPA9000 IP PBX
SPA400 SIP-PSTN Gateway and Voicemail Server
SPA900 Series IP Phones and Accessories
Deployment Scenarios
Deploying the LVS with PSTN Access and Local Voice Mail
Deploying the LVS with ITSP Service Only
Deploying the LVS with ITSP Service, PSTN Access and Local Voice Mail
ITSP Service, PSTN and ISDN Access and Local Voice Mail
Initial Installation, and Configuration

13
14
14
14
15
16
17
18
19
20

Chapter 2: Basic Administration of the SPA9000 . . . . . . . . . 21
Upgrading Firmware for the SPA9000
Connecting to the SPA9000 Administration Web Server
Saving or Discarding Changes SPA9000
Access Levels
Setting Passwords for User and Administrator Accounts
Configuring Basic Settings
Setting Up the WAN Connection for the SPA9000
Setting the Date and Time
Configuring Daylight Saving Time
SPA9000 Ethernet Port
LAN and Application Guidelines
Configuring Multicast Addressing and Group Paging
Collecting System Logs and Debug Information
Viewing Information about the SPA9000
Viewing Information about Client Stations
Using the Interactive Voice Response Unit
Using the IVR Menu
Entering a Password through the IVR

21
24
24
25
25
26
26
27
27
28
28
29
31
33
33
33
34
37

Chapter 3: Configuring the SPA9000 for ITSP Interoperability . . 39
About LVS and SIP
Network Address Translation (NAT) and Voice over IP (VoIP)
NAT Mapping with SIP-ALG Router
NAT Mapping with Session Border Controller
Linksys Voice System Administration Guide

39
40
40
41
i

Table of Contents

Configuring NAT Mapping with a Static IP Address
Configuring NAT Mapping with STUN
Firewalls and SIP
Configuring SIP Timer Values

41
42
43
44

Chapter 4: Configuring Phone Lines and Calling Routing Behavior 45
Configuring SPA9000 FXS Ports
45
Configuring Line Interfaces on the SPA9000
46
Configuring a Line Interface for ITSP Service
46
Configuring a Line Interface for a SPA400 (PSTN or Voice Mail)
48
Configuring Call Capacity for a Line Interface
51
Configuring Dial Plans
52
How the Dialed Digits are Processed
53
Digit Sequences
53
Digit Sequence Examples
55
Acceptance and Transmission the Dialed Digits
56
Editing the System Dial Plan
57
Entering a Phone Dial Plan
58
Entering the Line Interface Dial Plan
59
Setting the Timers for the Dial Plan
59
Managing the Line Selection for Outbound Calls
63
Line Availability
63
Configuring a Call Routing Rule
63
Entering a Call Routing Rule
65
Managing Caller ID Settings for Outgoing Calls
65
Managing Inbound Calls with the Contact List
66
Routing an Inbound Call to the Auto Attendant
66
Routing an Inbound Call to a Receptionist or Client Stations
66
Example Contact List Rules
66
Supporting Multiple DID Numbers Per Line Interface
68
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.
69
Supporting Direct Inward Dialing to Phone Extensions
69
Entering a Contact List Rule
70
Call Forwarding Support on SPA9000
71
Call Transfer Support on SPA9000
72
Call Forward Bridge Mode
72
Call Transfer Bridge Mode
72
Managing Inbound Calls with Hunt Groups
73
Hunt Group that Rings All Stations Simultaneously
73
Hunt Group that Rings Stations Individually
75
Managing Inbound Calls with Shared Line Appearances
77
About Shared Line Appearances
77

Chapter 5: Administering the SPA400 and Voice Mail Service . . 80
Connecting to the SPA400 Administration Web Server
Configuring the SPA400 Network Connection
Saving or Discarding Changes on the SPA400
Managing Access to the SPA400 Web-Based Configuration Utility
Upgrading the Firmware for the SPA400
Linksys Voice System Administration Guide

80
81
82
83
84
ii

Table of Contents

Configuring a SPA400 to Interoperate with the SPA9000
Configuring a SPA400 for PSTN Access
Configuring a SPA400 for Voice Mail Service
Voice Mail Capacity
Configuring Local Voice Mail Service on a SPA400
Setting Up Voice Mail on Each Station
Enabling Remote Voice Mail Access (Optional)
Managing the Voice Mail Messages on the USB Key
Enabling Debugging on the SPA400

85
87
87
88
88
90
93
94
95

Chapter 6: Configuring Music on Hold . . . . . . . . . . . . . . 97
Using the Internal Music Source for Music On Hold
Using the Internal Music Source
Changing the Music File for the Internal Music Source
Restoring the Original Music File
Configuring a Streaming Audio Server
Using a Streaming Audio Server
Using the IVR with an SAS Line
Example SAS with MOH
Configuring the Streaming Audio Server

97
97
98
98
99
99
100
100
101

Chapter 7: Configuring the LVS Auto-Attendant . . . . . . . . 103
How the Auto Attendant Works
Working with the Auto Attendant Greetings
Using Pre-Recorded Prompts
Recording an Auto Attendant Prompt
Downloading Prompts
Writing an Auto Attendant Script
An Introduction to XML Scripting Grammar in AA Script Examples
Elements of XML Scripting Grammar
Auto Attendant XML Instructions Set
Entering an Auto Attendant Script
Configuring the DayTime, NightTime and Weekend/Holiday Auto Attendants
Configuring Dial Plans for the Auto Attendant

103
104
104
105
107
108
109
111
113
115
116
117

Chapter 8: Localization . . . . . . . . . . . . . . . . . . . . 118
Requirements for Localization
Localizing the SPA9000 Auto Attendant Prompts
Local Time Configuration
Configuring the SPA9000 and SPA9xx Call Progress Tones
Localizing the SPA400 Voice Mail Prompts
Localizing the SPA400 Call Disconnect Tones
Localizing the SPA400 Caller ID Method

118
118
120
120
125
126
128

Appendix A: Advanced Topics in LVS Administration . . . . . . . . 130
Technology Background
Session Initiation Protocol
SPA9000 Media Proxy
Using the SPA9000 with a Firewall or Router
SPA400 SIP-PSTN Gateway
Linksys Voice System Administration Guide

130
130
132
133
133
iii

Table of Contents

SPA9000 Architecture
SIP-NAT Interoperation
Determining Whether the Router Uses Symmetric or Asymmetric NAT
Advanced Call Control and Routing
Configuring Vertical (Supplementary) Service Codes
Managing the Outbound Call Routing Groups
Configuring Outbound Call Codec Selection Codes
Advanced Topics for SPA400 Voice Mail Service
How Voicemail Works
Checking Voicemail from an External Number
Depositing Voicemail
Subscribing to Voicemail Notification
Remote Provisioning Features
Using Configuration Profiles
Client Auto-Configuration
Manual Client Configuration
Client Registration
Using the Upgrade URL
Using the Resync URL
Using the Reboot URL

134
135
136
137
137
139
141
141
142
142
143
144
145
145
146
147
149
150
151
151

Appendix B: SPA9000 Field Reference . . . . . . . . . . . . . . . 152
Router Tab
Status page
Wan Setup page
Lan Setup page and Application page
Voice tab
Info page
System page
SIP Page
Provisioning page
Regional page
FXS 1/2 page
Line 1/2/3/4 page

152
152
153
156
156
156
159
160
176
176
189
196

Appendix C: SPA400 Field Reference . . . . . . . . . . . . . . . . 203
Setup
Basic Setup
SPA9000 Interface
Voice
Voicemail Server
Voicemail Users
Administration
Management
Factory Default
USB Setting
Firmware Upgrade
Reboot
Status
Gateway
Event Logs
Linksys Voice System Administration Guide

203
203
204
205
208
209
209
209
210
210
210
210
211
211
212
iv

Table of Contents

Set Log Level
Tone

212
212

Appendix D: Warranty Information . . . . . . . . . . . . . . . . 214
Limited Warranty
Exclusions and Limitations
Obtaining Warranty Service
Technical Support

214
214
215
215

Appendix E: Regulatory Information . . . . . . . . . . . . . . . . 216
Federal Communications Commission Interference Statement
216
Industry Canada Statement
216
Règlement d’Industry Canada
216
EC Declaration of Conformity (Europe)
217
User Information for Consumer Products Covered by EU Directive 2002/96/EC on Waste
Electric and Electronic Equipment (WEEE)
217

Appendix F: Safety Information . . . . . . . . . . . . . . . . . . 224
Meaning of the Warning Symbol
General Safety Information
Power Safety Information

224
224
225

Appendix G: Software License Agreement . . . . . . . . . . . . . 227
Software in Linksys Products:
Software Licenses:
Schedule 1 Linksys Software License Agreement
Schedule 2
Schedule 3

227
227
227
229
234

Appendix H: Contacts . . . . . . . . . . . . . . . . . . . . . . . 237
Appendix I: Acronyms . . . . . . . . . . . . . . . . . . . . . . . 238
Appendix J: Glossary . . . . . . . . . . . . . . . . . . . . . . . . 241

Linksys Voice System Administration Guide

v

Document Audience

Preface
The Linksys Voice System Administration Guide is intended to help VARs and Service Providers to
manage and configure the Linksys Voice System (LVS). 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:
•

”Document Audience,” on page vi

•

”Document Purpose and Contents,” on page viii

•

”Document Conventions,” on page ix

•

”Related Documentation,” on page ix

•

”Finding Information in PDF Files,” on page xi

•

”Online Resources,” on page xiii

•

”Copyright and Trademarks,” on page xiii

Document Audience
This document is written for the following audience:
•

Service providers offering services using LVS products

•

VARs and resellers who need LVS configuration references

•

System administrators or anyone who performs LVS installation and administration

Note

This guide does not provide the configuration information required by
specific service providers. Please consult with the service provider for
specific service parameters.

Linksys Voice System Administration Guide

vi

Firmware

Firmware
This guide supports the following firmware releases. The installed firmware must be at least the
indicated in the table below.

Product

Firmware Version

SPA9000

5.1.9

SPA400

1.0.1.4

SPA922/SPA942

5.2.5

SPA962

5.2.8

SPA901

5.1.5

SPA921/SPA941

5.1.8

Linksys Voice System Administration Guide

vii

Document Purpose and Contents

Document Purpose and Contents
This document provides information that an administrator needs to configure the Linksys 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,” on page ix.
The information in this guide is organized into the following chapters and appendices:

Chapter 1, "Getting Started."

This chapter introduces you to the Linksys Voice System
(LVS) 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 the
SPA9000 for ITSP
Interoperability"

This chapter provides configuration details for the
purpose of helping you to ensure that your infrastructure
properly supports the LVS.

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 LVS
Auto-Attendant"

This chapter describes how to configure the LVS Auto
Attendant using the IVR, and XML scripting.

Chapter 8, "Localization"

You can localize your LVS with the language files, tones,
and ring patterns that are appropriate for your region.

Appendix A, "Advanced Topics
in LVS Administration"

This appendix provides technical information that is
useful to individuals who want a better understanding of
how the LVS works.

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 thes of
the SPA400 administration web server.

Linksys Voice System Administration Guide

viii

Document Conventions

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

Meaning

Boldface

May indicate either of the following:
• 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:
• 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.

Italic

Monospaced Font

Related Documentation
This LVS Administration Guide is part of a complete suite of documentation that is available to
assist you in using and configuring Linksys devices. The following documents are of special
interest to LVS administrators.

Note

These documents and more are available at Linksys.com.

Document Title

Description

Intended Audience

Linksys Voice System
Installation and
Configuration Guide

•

Network design
considerations and site
preparation
Switch configuration
Initial installation and
configuration of the LVS
components
SPA9000, SPA400,
SPA900 series IP phones

VARs and Service Providers

Configuration and
management of IP
phones
Deployment options
with or without the
SPA9000 IP PBX
SPA9x2 series IP phones

VARs and Service Providers

•
•

•
Linksys Phone
Administration Guide

•

•

•

Linksys Voice System Administration Guide

ix

Related Documentation

Document Title

Description

Intended Audience

Linksys SPA9x2 Phone User
Guide

•
•
•

Phone setup
Phone features
SPA9x2 series IP phones

VARS and phone end-users

Linksys Analog Telephone
Adapter Administration
Guide

•

Administration and use
of Linksys ATAs
PAP2T, SPA2102,
SPA3102, SPA8000,
AG310, RTP300, WRP400,
and WRTP54G

VARS, system administrators,
and Service Providers

Linksys Provisioning Guide

•

Provisioning LVS
components

Service Providers only

SPA9000 FAQS Specification, Troubleshooting,Configuration, and General
Specifications

Answers to Frequently Asked
Questions, within the Easy
Answers Knowledge Base at
www.linksys.com/kb

•

Linksys Voice System Administration Guide

VARs and Service Providers

x

Finding Information in PDF Files

Finding Information in PDF Files
The PDF Find/Search tool lets you find information quickly and easily online. You can:
•

Search an individual PDF.

•

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

•

Perform advanced searches.

Finding Text in a PDF
1. Enter your search terms in the Find 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.

2. Optionally, click the arrow next to the Find text box to refine your search by choosing
special options such as Whole words only.
3. Press Enter. Acrobat displays the first instance of the search term. Press Enter again to
continue to more instances of the term.

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.
1. Start Acrobat Professional or Adobe Reader.
2. Choose Edit > Search, or click the arrow next to the Find box and then choose Open Full
Acrobat Search.

3. In the Search window, complete the following steps:
a. Enter the text that you want to find.
b. Choose All PDF Documents in.
c. From the drop-down box, choose Browse for Location. Then choose the location on
your computer or local network, and click OK.
Linksys Voice System Administration Guide

xi

Finding Information in PDF Files

d. If you want to specify additional search criteria, click Use Advanced Search Options,
and choose the options you want.
e. Click Search.

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

Note

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

Linksys Voice System Administration Guide

xii

Online Resources

Online Resources
Website addresses in this document are listed without http:// in front of the address because
most current web browsers do not require it. If you use an older web browser, you may have to
add http:// in front of the web address.

Resource

Link

Linksys

www.linksys.com

Linksys International

www.linksys.com/international

Glossary

www.linksys.com/glossary

Network Security

www.linksys.com/security

Copyright and Trademarks
Linksys is a registered trademark or trademark of
Cisco Systems, Inc. and/or its affiliates in the U.S.
and certain other countries. Copyright © 2008
Cisco Systems, Inc. All rights reserved. Other
brands and product names are trademarks or
registered trademarks of their respective
holders.

Technical Support
A list of technical support phone numbers and websites is available in Appendix H, "Contacts."

Linksys Voice System Administration Guide

xiii

Getting Started

1

Introduction to the Linksys Voice System

Getting Started
This chapter introduces you to the Linksys Voice System (LVS) by describing the components
and presenting several deployment scenarios.
•

”Introduction to the Linksys Voice System,” on page 13

•

”Deployment Scenarios,” on page 15

•

”Initial Installation, and Configuration,” on page 20

Introduction to the Linksys Voice System
The LVS is an affordable and feature-rich voice over IP (VoIP) telephone system that is designed
especially for small businesses. The LVS uses standard TCP/IP protocols and can provide global
connectivity through any Internet Telephony Service Provider (ITSP) that supports Session
Initiation Protocol (SIP).
At minimum, the LVS 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 LVS can subscribe to ITSP services to take advantage of low calling rates. With
the optional SPA400, the LVS can connect to the Public Switched Telephone Network (PSTN) to
support legacy phone lines. Figure 1-1 illustrates the LVS Solution in a typical deployment.
Figure 1-1 The Linksys Voice System (LVS) with the SPA9000 and SPA400

SPA9xx IP Phones

SPA400
for Voice Mail

SLM224P Switch

SPA9000 IP PBX
Analog Phones
or Fax Machines

SPA400
for PSTN Access
Analog
Phone Lines

WRV200 Router

Integrated
Access Device

Internet

Linksys Voice System Administration Guide

13

Getting Started
Introduction to the Linksys Voice System

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 licence's 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.
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 input (key presses that produce DTMF tones) following the rules
specified in the Auto Attendant script on the SPA9000.

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
LVS users 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.

SPA900 Series IP Phones and Accessories
Linksys provides a variety of IP phones to meet the needs of small businesses. Table 1-1
provides a comparison of the available models.
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 Linksys Phone
Administration Guide and the Linksys SPA9x2 Phone User Guide.
Table 1-1 Linksys SP900 Series IP Phones and Accessories

Product
SPA922*

Linksys Voice System Administration Guide

RJ-45
2

Voice Lines
1

Additional Features/
Notes
One-line IP phone with
Power over Ethernet (PoE)
support

14

Getting Started
Deployment Scenarios

Product

RJ-45

Voice Lines

Additional Features/
Notes

SPA932

—

—

Attendant console (sidecar)
for SPA962 with 32 buttons
and LEDs for monitoring
and call transfer

SPA942*

2

4

Four-line IP phone with
Power over Ethernet (PoE)
support

SPA962*

2

6

Six-line IP Phone with highresolution color display
and Power over Ethernet
(PoE) support

POES5

1

N/A

Provides an 802.3af PoE
port for connection back to
a PoE switch for SPA9000,
SPA400 and SPA9x1
phones

WBP54G

1

N/A

Converts your IP phone
into a wireless device, so it
can connect to your
wireless network without
an Ethernet cable

* NOTE: SPA922, SPA942, SPA962 do not include an external power adapter as they support
Power over Ethernet (802.3af ). If the phones are connected to a non-POE switch, the PA100
power adapter is required.

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

”Deploying the LVS with PSTN Access and Local Voice Mail,” on page 16

•

”Deploying the LVS with ITSP Service Only,” on page 17

•

”Deploying the LVS with ITSP Service, PSTN Access and Local Voice Mail,” on page 18

Linksys Voice System Administration Guide

15

Getting Started
Deployment Scenarios

•

”ITSP Service, PSTN and ISDN Access and Local Voice Mail,” on page 19

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

SPA9xx IP Phones

SPA400
for Voice Mail

SLM224P Switch

SPA9000 IP PBX

SPA400
for PSTN Access

Analog
Phone Lines

Linksys Voice System Administration Guide

16

Getting Started
Deployment Scenarios

Deploying the LVS with ITSP Service Only
In this scenario, a customer has no legacy telephone numbers and does not require local voice
mail (either needs no voice mail at all or has voice mail hosted by the ITSP). The LVS is deployed
with the SPA9000 IP PBX, VoIP service, and four SPA9x2 series phones. Four phones are
connected at this time, but the LVS can be expanded to include 16 SPA9xx IP phones.
Optionally, analog phones or fax machines (not illustrated) can be connected to the two phone
ports on the SPA9000.

SPA9xx IP Phones

SLM224P Switch
SPA9000 IP PBX

WRV200 Router

Integrated
Access Device

Internet

Linksys Voice System Administration Guide

17

Getting Started
Deployment Scenarios

Deploying the LVS with 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 over the
original legacy telephone number, or to route local telephone calls). This customer also prefers
local voice mail service. The LVS is deployed with the SPA9000 IP PBX, VoIP service, one SPA400
unit for voice mail service, another SPA400 units for PSTN access with four FXO ports, and up to
16 SPA900 series phones. Optionally, analog phones or fax machines (not illustrated) can be
connected to the two phone ports on the SPA9000.

SPA9xx IP Phones

SPA400
for Voice Mail

SLM224P Switch

SPA9000 IP PBX

SPA400
for PSTN Access
Analog
Phone Lines

WRV200 Router

Integrated
Access Device

Internet

Linksys Voice System Administration Guide

18

Getting Started
Deployment Scenarios

ITSP Service, PSTN and ISDN Access and Local Voice Mail
In this scenario, the customer takes full advantage of the LVS solution, with the SPA9000 IP PBX,
VoIP service, one SPA400 unit for voice mail service, another SPA400 for PSTN access with four
FXO ports, one ISDN Gateway for ISDN BRI access with four BRI ports, and up to 16 SPA9xx IP
phones. Optionally, analog phones or fax machines (not illustrated) can be connected to the
two phone ports on the SPA9000.

SPA9xx IP Phones

SPA400
for Voice Mail

SLM224P Switch

SPA9000 IP PBX

SPA400
for PSTN Access
WRV200 Router

Analog
Phone Lines
Mediatrix® 4402
BRI Gateway
ISDN
Phone Lines

Linksys Voice System Administration Guide

Integrated
Access Device

Internet

19

Getting Started
Initial Installation, and Configuration

Initial Installation, and Configuration
Linksys strongly recommends that you use the Linksys 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 Linksys 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.

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Basic Administration of the SPA9000

2

Upgrading Firmware for the SPA9000

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.
For Installation and Configuration of LVS in the most common deployment scenarios, refer to
the Linksys Voice System Installation and Configuration Guide, which is available at Linksys.com.
See the following topics:
•

”Upgrading Firmware for the SPA9000,” on page 21

•

”Connecting to the SPA9000 Administration Web Server,” on page 24

•

”Saving or Discarding Changes SPA9000,” on page 24

•

”Access Levels,” on page 25

•

”Setting Passwords for User and Administrator Accounts,” on page 25

•

”Configuring Basic Settings,” on page 26

•

”Viewing Information about the SPA9000,” on page 33

•

”Viewing Information about Client Stations,” on page 33

•

”Using the Interactive Voice Response Unit,” on page 33

Upgrading Firmware for the SPA9000
As needed, you can download new firmware from Linksys.com and then install the firmware on
the SPA9000.
1. Download the firmware from Linksys.com by completing the following steps:
a. Start a web browser and enter the following address: www.linksys.com.
b. On the menu at the top of the page, choose Support > Technical Support.
c. Click Choose a Product.
d. From the Voice over IP (VoIP) drop-down list, choose IP PBX.
e. On the Choose the Device IP PBX page, choose SPA9000 from the drop-down list on the
left side of the page.

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Basic Administration of the SPA9000
Upgrading Firmware for the SPA9000

f.

On the Technical Support page, click the link in the Downloads section.

g. On the SPA9000 Downloads page, choose the version from the Please select a version
drop-down list.
h. In the Firmware section, click the link for the version that you want to install, and then
save the file on the administration computer.
2. Extract the Zip file, and then run the executable file to upgrade the firmware. When the
Firmware Upgrade Warning window appears, click Continue.

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

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

5. When the confirmation message appears, click OK.

6. To verify the upgrade, complete the following steps:

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

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 24).
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.

Connecting to the SPA9000 Administration Web Server
To connect to the SPA9000 administration web server, perform the following steps.
1. Start Internet Explorer on any computer that is connected to the same network as the
SPA9000.
NOTE: You should have previously connected and configured the SPA9000 as described in
the LVS Installation Guide.
2. Enter the IP address of the SPA9000.
NOTE: You can find the IP address of the SPA9000 by connecting 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.
3. To view administrative features, click Admin Login and then click Advanced.
NOTE: By default, no password is required. For more information, see ”Setting Passwords for
User and Administrator Accounts,” on page 25.
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.

•

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.

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Basic Administration of the SPA9000
Access Levels

TIP: Before you make changes, it is recommended
that you save a copy of your current working
configuration:
1. In Internet Explorer, connect to the
administration web server.
2. From the menu, choose File > Save As.
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.
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 (with a lower-case a) and the User
account name is user. These account names 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.

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Basic Administration of the SPA9000
Configuring Basic Settings

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 24.)
2. Click Voice tab > System.
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.
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 26

•

”Setting the Date and Time,” on page 27

•

”Configuring Daylight Saving Time,” on page 27

•

”LAN and Application Guidelines,” on page 28

•

”Configuring Multicast Addressing and Group Paging,” on page 29

•

”Collecting System Logs and Debug Information,” on page 31

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.
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 24.)
2. Click Router tab > Wan Setup.
3. From the Connection Type drop-down list, choose Static IP.
4. In the Static IP Settings area, enter the Static IP of the SPA9000, as well as the NetMask and
Gateway for your network.
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.
6. Click Submit All Changes. The SPA9000 reboots.

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Basic Administration of the SPA9000
Configuring Basic Settings

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.

Note

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

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 24).
2. Click Router tab > Wan Setup.
3. Scroll down to the Optional settings section.
4. Enter the fully qualified domain name of the NTP server that you want to use, such as
time.nist.gov.
5. Click Submit All Changes.
6. Click Voice tab > Regional.
7. Scroll down to the Miscellaneous section.
8. From the Time Zone drop-down list, choose your local time zone.
9. Click Submit All Changes.
10.

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  and  values specify the start and end dates and times of
daylight saving time. Each value is in the following format:  / /
[/HH:[mm[:ss]]]

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Basic Administration of the SPA9000
Configuring Basic Settings

•

The  value is the number of hours, minutes, and/or seconds to add to the
current time during daylight saving time. The  value can be preceded by a
negative (-) sign if subtraction is desired instead of addition. The  value is in
this format: [/[+|-]HH:[mm[:ss]]]

•

The  value equals any value in the range 1-12 (January-December).

•

The  value equals [+|-] any value in the range 1-31. If  is 1, it means the
 on or before the end of the month (in other words the last occurrence of
 in that month).

•

The  value equals any value in the range 1-7 (Monday-Sunday). It can also
equal 0. If the  value is 0, this means that the date to start or end daylight
saving is exactly the date given. In that case, the  value must not be negative. If
the  value is not 0 and the  value is positive, then daylight saving
starts or ends on the  value on or after the date given. If the 
value is not 0 and the  value is negative, then daylight saving starts or ends on the
 value on or before the date given.

•

The abbreviation HH stands for hours (0-23).

•

The abbreviation mm stands for minutes (0-59).

•

The abbreviation ss stands for seconds (0-59).

Entering the Daylight Saving Time Rule
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 24).
2. Click Voice tab > Regional.
3. Scroll down to the Miscellaneous section.
4. Enter the rule in the Daylight Saving Time Rule field.
5. Click Submit All Changes.

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.
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Basic Administration of the SPA9000
Configuring Basic Settings

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 LVS
Installation and Configuration Guide.

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

Graceful reboot

•

Immediate reboot

•

Graceful restart

•

Immediate restart

•

Group page start

•

Group page end

•

Get ringing calls

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 24).
2. Click Voice tab > SIP.
3. Scroll down to the PBX Parameters section.
4. Enter the correct multicast address in the Multicast Address field.

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Basic Administration of the SPA9000
Configuring Basic Settings

NOTE: The default value is 224.168.168.168:6061.
5. Click Submit All Changes.
6. Enter the same multicast address in the phone configurations:
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.
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 24).
2. Click Voice tab > SIP.
3. Scroll down to the PBX Parameters section.

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Basic Administration of the SPA9000
Configuring Basic Settings

4. Enter the correct multicast address in the Group Page Address field.

NOTE: The default value is 224.168.168.168:34567.
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.

•

If you are interested in working with these logs yourself, you will need to use an
application that reads syslogs. Linksys partners can download a syslog server tool
(slogsrv.exe) from the Linksys Partner Connection. (VARs, see Product Utilities > Linksys
Voice System > SPA Utilities > Syslog Server for SPA Devices. SPs, see Technical Tools >
SPA Utilities > Syslog Server for SPA Devices.)

Note

As a best practice, enable logging only when needed, and disable
logging when you finish the investigation. Logging information can
impact system performance.

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 24).
2. Click Voice tab > System.

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Configuring Basic Settings

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.

i.

Debug Level: Choose 3 to enable debugging.

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

•

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, NTFY, and REG
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, NTFY, and REG
information to reduce the length of the logs.

EXAMPLES:
•

If you are troubleshooting a problem with line registration. Select full.

•

If you are troubleshooting a call problem. Select full excl. OPT|NTFY|REG.

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.
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.
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 24).
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

3. 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.
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Basic Administration of the SPA9000
Using the Interactive Voice Response Unit

•

”Using the IVR Menu,” on page 34

•

”Entering a Password through the IVR,” on page 37

Using the IVR Menu
To use the IVR menu, complete the following steps.
1. Connect an analog telephone to the Phone 1 or Phone 2 port of the SPA9000.
2. Press **** (quickly press the star key four times).
3. Wait until you hear “Linksys configuration menu.”
4. Refer to Table 2-1 to identify the option required.
5. Enter the required option followed by the # (pound) key.
NOTE:
•

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).
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.

•

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

6. 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.

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Basic Administration of the SPA9000
Using the Interactive Voice Response Unit

Table 2-1 IVR Options

IVR Action

IVR Menu
Choice

Parameter(s)

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
Check DHCP

3948
100

None
None

Enable/Disable
DHCP

101

Enter 0 to enable

Check WAN IP
Address
Set Static IP
Address

110

Enter 1 to disable
None

111

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

Check Network
Mask
Set Network Mask

120

None

121

Enter value using
numbers on the
telephone key pad. Use
the * (star) key when
entering a decimal
point.
None

Check Static
Gateway IP
Address

130

Linksys Voice System Administration Guide

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.
Requires password

IVR announces the current IP
address of the WAN port.
DHCP must be “Disabled,”
otherwise you hear, “Invalid
Option,” if you try to set this
value. 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
IVR announces the current
network mask of SPA.
DHCP must be “Disabled,”
otherwise you hear, “Invalid
Option,” if you try to set this
value.
Requires password
IVR announces the current
gateway IP address of SPA.

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Basic Administration of the SPA9000
Using the Interactive Voice Response Unit

Table 2-1 IVR Options (Continued)

Set Static Gateway
IP Address

Check MAC
Address
Check Firmware
Version
Check Primary
DNS Server
Setting
Set Primary DNS
Server

Check
administration
web server port
Check LAN IP
Address
Check PBX
multicast address
Set PBX multicast
address

Enable/Disable
administration
web server
Manual Reboot of
Unit

131

140

Enter IP address using
numbers on the
telephone key pad. Use
the * (star) key when
entering a decimal
point.
None

150

None

160

None

161

170

Enter IP address using
numbers on the
telephone key pad. Use
the * (star) key when
entering a decimal
point.
None

210

None

180

None

181

Enter IP address and
port. Use * key for
entering a dot. For
example,
224.168.168.169:8089 is
224*168*168*169*8089.
Enter 1 to enable
Enter 0 to disable

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

None

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

7932

732668

Linksys Voice System Administration Guide

DHCP must be “Disable,”
otherwise you hear, “Invalid
Option,” if you try to set this
value.
Requires password
IVR announces the MAC address
of SPA in hex string format.
IVR announces the version of the
firmware running on the SPA.
IVR announces the current
setting in the 
parameter.
Requires password

IVR announces the port that the
web server is listening on.
(Default is 80)
IVR announces the current IP
address of the LAN port.
IVR announces the current value.

Requires password

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Basic Administration of the SPA9000
Using the Interactive Voice Response Unit

Table 2-1 IVR Options (Continued)

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 NON-DEFAULT
SETTINGS WILL BE
LOST!
This includes
network and
service provider
data.

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’

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–

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’

Note

This translation convention only applies to the password input.

For example, to input password test#@1234 by phone keypad, you need to press the
following sequence of digits: 8378001234.
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).

NOTES:
–

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.

2. 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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Configuring the SPA9000 for ITSP Interoperability

3

About LVS and SIP

Configuring the SPA9000 for ITSP
Interoperability
This chapter provides configuration details for the purpose of helping you to ensure that your
infrastructure properly supports the LVS.
•

”About LVS and SIP,” on page 39

•

”Network Address Translation (NAT) and Voice over IP (VoIP),” on page 40

•

”Firewalls and SIP,” on page 43

•

”Configuring SIP Timer Values,” on page 44

About LVS and SIP
The LVS is implemented using open standards, such as Session Initiation Protocol (SIP), to help
ensure interoperation 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. Figure 3-1 illustrates the SIP requests and responses between client
stations in the LVS. 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.
Figure 3-1 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 3-2, 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.
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Network Address Translation (NAT) and Voice over IP (VoIP)

Figure 3-2 SPA9000 as a SIP Proxy for Internet Calls

UserC

UserA

UserB
Hub/switch

IP Router (firewall)
Broadband modem
ISP

Internet

Internet (WAN)
Interface

SPA9000
ITSP
SIP Proxy with
media proxy enabled

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 SIP-ALG Router,” on page 40

•

”NAT Mapping with Session Border Controller,” on page 41

•

”Configuring NAT Mapping with a Static IP Address,” on page 41

•

”Configuring NAT Mapping with STUN,” on page 42

NAT Mapping with SIP-ALG Router
You can achieve NAT mapping by using a router that has a SIP ALG (Application Layer Gateway).
The Linksys WRV200 router is recommended for this purpose, although any router with a SIPALG can be used. By using a SIP-ALG router, you have more choices in selecting an ITSP.

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Configuring the SPA9000 for ITSP Interoperability
Network Address Translation (NAT) and Voice over IP (VoIP)

NAT Mapping with Session Border Controller
It is strongly recommended that you choose an ITSP that supports NAT mapping through a
Session Border Controller (see Figure 3-3). With NAT mapping provided by the ITSP, you have
more choices in selecting a router.
Figure 3-3 NAT Support with Session Border Controller Provided by ITSP
Private IP address
192.168.1.1
192.168.1.101

External IP address
assigned by ISP

192.168.1.102
NAT Device
Internet

ISP

Router and
DHCP
DHCP Server
server

SPA9000
ITSP
SIP Proxy
192.168.1.100

Session Border
Controller

Configuring NAT Mapping with a Static IP Address
Configuring NAT mapping in the SPA9000 is recommended only if the ITSP network does not
provide a Session Border Controller functionality. In this case, and if the external (public) IP
address is static, then it is recommended to perform a static (permanent) mapping on SPA9000.
This setting is also recommended if NAT mechanism used in the router is symmetric.

Note

The LAN switch needs to be configured to enable Spanning Tree
Protocol and Port Fast on the ports to which the SPA9000 and the
SPA9xx phones are connected.

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 24.)
2. Click Voice tab > SIP.
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.

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

SPA9000 Voice > SIP: NAT Support Parameters

4. Click Voice tab > Line N, where N is the number of the line interface for the ITSP service that
you need to configure.
5. Scroll down to the NAT Settings section.
•

NAT Mapping Enable: Choose YES.

•

NAT Keep Alive Enable: Choose YES (optional).
NOTE: Your ITSP may require the SPA9000 to send NAT keep alive messages to keep the
NAT ports open permanently. For more information, see Appendix A, "Advanced Topics
in LVS Administration," ”SIP-NAT Interoperation” section on page 135.

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 43.

Configuring NAT Mapping with STUN
Configuring NAT mapping in the SPA9000 is recommended only if the ITSP network does not
provide a Session Border Controller functionality. In this case, and if the external (Public) IP
address is assigned dynamically by the network (and the router uses asymmetric NAT
mechanism), it is possible to use STUN as a mechanism to discover the NAT mapping in
SPA9000. This is considered a practice of last resort and should be used only if the other
methods are unavailable.

Note

STUN is a viable option only if your router uses asymmetric NAT. See
Appendix A, "Advanced Topics in LVS Administration," ”Determining
Whether the Router Uses Symmetric or Asymmetric NAT” section on
page 136.

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 24.)
2. Click Voice tab > SIP.

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Firewalls and SIP

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, Insert VIA received, Substitute VIA Addr: yes

•

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

•

STUN Enable: Choose yes.

•

STUN Server: Enter the IP address for your STUN server.

4. Click Voice tab > Line N, where N is the number of the line interface for the ITSP service that
you need to configure.
5. Scroll down to the NAT Settings section.
•

NAT Mapping Enable: Choose YES.

•

NAT Keep Alive Enable: Choose YES (optional).
NOTE: Your ITSP may require the SPA9000 to send NAT keep alive messages to keep the
NAT ports open permanently. For more information, see Appendix A, "Advanced Topics
in LVS Administration," ”SIP-NAT Interoperation” section on page 135.

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 43.

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 SPA9000 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

Configuring SIP Timer Values
The SPA9000 is configured with default timer values that 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. For field descriptions, see ”SIP Timer Values (sec) section,” on
page 161.

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4

Configuring Phone Lines and Calling Routing Behavior
Configuring SPA9000 FXS Ports

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 45

•

”Configuring Line Interfaces on the SPA9000,” on page 46

•

”Configuring Dial Plans,” on page 52

•

”Managing the Line Selection for Outbound Calls,” on page 63

•

”Managing Caller ID Settings for Outgoing Calls,” on page 65

•

”Managing Inbound Calls with the Contact List,” on page 66

•

”Managing Inbound Calls with Hunt Groups,” on page 73

•

”Managing Inbound Calls with Shared Line Appearances,” on page 77

Configuring SPA9000 FXS Ports
The SPA9000 FXS ports can be used to connect analog phones and fax machines to the LVS. 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

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 24.)
2. Click Voice tab > FXS N, where N is the port number.
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.

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Configuring Line Interfaces on the SPA9000

•

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).

4. Enter the Dial Plan settings, as needed. See ”Configuring Dial Plans,” on page 52.
5. Click Submit All Changes.

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, available at www.linksys-itsp.com
(for Europe, Middle East and Africa Service Providers) and www.linksys-voip.eu (for
Europe, Middle East and Africa Value Added Resellers).

This section includes the following topics:
•

”Configuring a Line Interface for ITSP Service,” on page 46

•

”Configuring a Line Interface for a SPA400 (PSTN or Voice Mail),” on page 48

•

”Configuring Call Capacity for a Line Interface,” on page 51

Configuring a Line Interface for ITSP Service
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 24.)
2. Click Voice tab > Line N, where N represents the line interface number.
3. From the Line Enable drop-down list, choose yes.
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)
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Configuring Line Interfaces on the SPA9000

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

NOTES:

•

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 66.

•

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.

5. Click the Submit All Changes button. The SPA9000 device reboots.
6. 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.

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Configuring Line Interfaces on the SPA9000

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.

Note

The LVS can be configured to support multiple DIDs per Line Interface.
For more information, ”Supporting Direct Inward Dialing to Phone
Extensions,” on page 69.

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.
NOTE:
•

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

•

Before you begin this procedure, connect the SPA400 as described in either the SPA400
Quick Install Guide or the LVS Installation and Configuration Guide.

•

Determine the DHCP-obtained IP address of the SPA400 by reviewing 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.

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 24.)
2. Click Voice tab > Line N, where N represents the line interface number.
3. From the Line Enable drop-down list, choose yes.
4. Enter the following information:
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Configuring Line Interfaces on the SPA9000

•

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 page 85.

•

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

5. 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).

•

Set XFER Bridge Mode to all.

•

Set CFWD Bridge Mode to all.

NOTES:

•

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

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Configuring Line Interfaces on the SPA9000

contact list. For more information, see ”Managing Inbound Calls with the Contact List,”
on page 66.
6. 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 page 85 and ”Configuring
a SPA400 for PSTN Access,” on page 87.

7. 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.
SPA9000 Voice > Line

8. Click the Submit All Changes button. 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 page 88.

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Configuring Line Interfaces on the SPA9000

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 51

•

”Setting the Call Capacity Parameter,” on page 52

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 LVS 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.
Table 4-1 provides the approximate bandwidth budget for different codecs.

Table 4-1 Bandwidth Budget

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

Codec

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Configuring Dial Plans

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.
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 24.)
2. Click Voice tab > Line N, where N represents the line interface number.
3. Scroll down to the Subscriber Information section.
4. From the Call Capacity drop-down list, choose the maximum number of calls to allow.
SPA9000 Voice > Line: Subscriber Information

5. Click Submit All Changes.

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:
•

”How the Dialed Digits are Processed,” on page 53

•

”Digit Sequences,” on page 53

•

”Digit Sequence Examples,” on page 55

•

”Acceptance and Transmission the Dialed Digits,” on page 56

•

”Editing the System Dial Plan,” on page 57

•

”Entering a Phone Dial Plan,” on page 58

•

”Setting the Timers for the Dial Plan,” on page 59

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How the Dialed Digits are Processed
When a user lifts a handset or presses a speaker button, the following sequence of events
begins:
1. The phone begins collecting the dialed digits. The inter-digit timers starts tracking the time
that elapses between digits. For more information, see ”Setting the Timers for the Dial Plan,”
on page 59.
2. If the inter-digit timer value is reached, or if another terminating event occurs, the phone
compares the dialed digits with the PHONE (SYSTEM) dial plan (on the SPA9xx Telephone
Configuration > Ext N page, Dial Plan section). For more information, see ”Acceptance and
Transmission the Dialed Digits,” on page 56, ”Editing the System Dial Plan,” on page 57, and
”Entering a Phone Dial Plan,” on page 58.
NOTE: If a call is placed from a phone that is connected to a SPA9000 FXS port, the FXS port
dial plan applies. See ”Configuring a Line Interface for a SPA400 (PSTN or Voice Mail),” on
page 48.
3. If the phone dial plan allows the call to process, the dialed numbers are sent to the SPA9000.
4. The SPA9000 compares the dialed digits to the CALL ROUTING RULE (on the SPA9000 Voice >
SIP page, PBX Parameters section). For more information, see ”Configuring a Call Routing
Rule,” on page 63.
5. If the call routing rule allows the call to process, then the SPA9000 compares the dialed
digits to the LINE INTERFACE dial plan (on the SPA9000 Voice > Line N page, Dial Plan
section). For more information, see ”Entering the Line Interface Dial Plan,” on page 59.
6. The SPA9000 uses the information in the line dial plan to manipulate the number (for
example, to remove steering digits) and then transmits the number.

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. The sequence can include the elements that are listed in Table 4-2.

Note

White space is ignored, but may be used for readability.
Table 4-2 Digit Sequences and Functions

Digit Sequence

Function

01234567890*#

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.

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Configuring Dial Plans

Digit Sequence

Function

[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.
• 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 *.

.
(period)

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 seven-digit 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

,
(comma)

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

S0

Enter S0 to reduce the timer to 0 so that the dialed digits are
evaluated after 0 seconds.
For more information about modifying the timers for the
evaluation of dialed digits, see ”Setting the Timers for the
Dial Plan,” on page 59.

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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

•

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 10-digit 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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•

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.

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 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.

•

•

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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.

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Configuring Dial Plans

Terminating Event

Processing

A timeout occurs.

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.
For more information, see ”Setting the
Timers for the Dial Plan,” on page 59.

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

•

•

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.

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.
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 24.)
2. Click Voice tab > SIP.
3. Scroll down to the PBX Phone Parameters section.
SPA9000 Voice > SIP: PBX Phone Parameters

4. Enter the digit sequences in the Dial Plan field. For more information and examples, see
”Digit Sequences and Functions,” on page 53.

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Configuring Dial Plans

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)

5. Click Submit All Changes. The phones reboot.
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 53.

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).

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 24.)
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.
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.
4. Click the Ext 1 tab, or the tab for the extension that you want to configure.
5. Scroll down to the Dial Plan section.
6. Enter the digit sequences in the Dial Plan field.
NOTE:
•

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

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Configuring Dial Plans

dial plan. For more information and examples, see ”Digit Sequences and Functions,” on
page 53.
•

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)

7. Click Submit All Changes. The phone reboots.
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.
9. Click the browser’s Back button to return to the list of phones.
10. Repeat this procedure for each client station that needs a unique dial plan.
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 53.

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.
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 24.)
2. Click Voice tab > Line N, where N represents the line interface number.
3. Scroll down to the Dial Plan section.
4. Enter the digit sequences in the Dial Plan field.
5. Click Submit All Changes.

Setting the Timers for the Dial Plan
The system uses dial plan timers to determine when a user’s entry (or a null entry) is evaluated.
You can modify the timers for an entire dial plan or for individual sequences within a dial plan.
There are three types of control timers:
•

”Dial Plan Timer (Off-Hook Timer),” on page 60

•

”Interdigit Long Timer (Incomplete Entry Timer),” on page 61

•

”Interdigit Short Timer (Complete Entry Timer),” on page 61

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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[2-9]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.

Setting the Dial Plan Timer for the System Dial Plan

Follow the procedure ”Editing the System Dial Plan,” on page 57.

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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 ”Editing the Control Timers for the
Default Interdigit Values,” on page 62.

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.
Setting the Interdigit Long Timer for the System Dial Plan

Follow the procedure ”Editing the System Dial Plan,” on page 57.
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.

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 ”Editing the Control Timers for the
Default Interdigit Values,” on page 62.

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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.
Setting the Interdigit Long Timer for the System Dial Plan

Follow the procedure ”Editing the System Dial Plan,” on page 57.
Setting the Interdigit Long Timer for the Phone Dial Plan

Follow the procedure ”Entering a Phone Dial Plan,” on page 58.
Editing the Control Timers for the Default Interdigit Values

Note

Use the following procedure to edit the default timer settings for all
calls. 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 Table 4-2, ‘Digit
Sequences and Functions,” on page 53.

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 24.)
2. Click Voice tab > Regional.
3. Scroll down to the Control Timer Values section.
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Managing the Line Selection for Outbound Calls

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.
SPA9000 Voice > Regional: Control Timer Values

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 section includes the following topics:
•

”Line Availability,” on page 63

•

”Configuring a Call Routing Rule,” on page 63

•

”Entering a Call Routing Rule,” on page 65

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 46.

•

The line is authorized by the Call Routing Rule for the dialed number. See”Configuring a
Call Routing Rule,” on page 63.

•

The line has capacity to take more calls. See ”Configuring Call Capacity for a Line
Interface,” on page 51.

•

The Dial Plan for this line allows the dialed number. See ”Configuring Dial Plans,” on
page 52.

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”

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•

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

•

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 Table 4-2, ‘Digit Sequences and Functions,” on page 53.

Any line interface can be used for allowed digit sequences
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 Table 4-2, ‘Digit Sequences and Functions,” on
page 63.

•

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).

•

Wildcard characters
EXAMPLE: (<:L1>51*,577?)

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Managing Caller ID Settings for Outgoing Calls

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
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 24.)
2. Click Voice tab > SIP.
3. Scroll down to the PBX Parameters section.
4. In the Call Routing Rule field, enter the rule that you want to apply
5. Click Submit All Changes.
6. To verify your progress, place a call to a phone that has caller ID, and confirm that the
expected number appears.

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 68.)

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.

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 24.)
2. Click Voice > Line N, where N represents the line interface number.
3. Scroll down to the Subscriber Information section.
4. Enter the desired display name for Caller ID in the Display Name field.
5. Click Submit All Changes.

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Managing Inbound Calls with the Contact List

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 66

•

”Routing an Inbound Call to a Receptionist or Client Stations,” on page 66

•

”Example Contact List Rules,” on page 66

•

”Entering a Contact List Rule,” on page 70

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 LVS 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
Line 1 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.

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

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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 972555-0156 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.

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 408555-0123 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 by ringing stations randomly
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.
SYNTAX: hunt=algo;interval;max, cfwd=target
–

algo: The order to ring the stations
restart (re)—Always start from beginning of the list
next (ne)—Start from the next on the list to the last station that rings

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random (ra)—Order is randomized for each call
–

interval: The number of seconds to ring each station

–

max: The total time in seconds or the total number of cycles to elapse before the call
is rejected or forwarded to the target. If max is less than interval, it is interpreted as
the number of cycles. If max is 0, hunting goes on indefinitely until the caller hangs
up or some one answers the call.

NOTE: You also can create a hunt group that acts as a separate extension within the
system. For more information, see ”Managing Inbound Calls with Hunt Groups,” on
page 73.

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.
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|…|4089993009:3009

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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 LVS
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 4-3 provides an example.
Table 4-3 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

NOTE:
•

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:
:+|:+|:+|:+|:
+|:+|:+|:+|:+|
:+|:+|:+|:+|
:+|:+|:+|

EXAMPLE:
4085555550:aa|4085555551:+101|4085555552:+102|4085555553:+103|4085555554:+104|408555565
0:+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.

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–

-  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 user-ID 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.
NOTE: 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|4085555553:+103|4085555554:
+104|4085555650:+500|

Entering a Contact List Rule

Note

The maximum length of the  parameter is 383 characters.

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 24.)
2. Click Voice tab > Line N, where N represents the line interface number.
3. Scroll down to the Subscriber Information section.
SPA9000 Voice > Line: Subscriber Information

4. Enter the desired rules in the Contact List field. For more information, see ”Example Contact
List Rules,” on page 66 and ”Entering a Contact List Rule,” on page 70.

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Call Forwarding Support on SPA9000

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.
6. Click Submit All Changes. The SPA9000 reboots.
7. To verify your progress, make an inbound call and verify that the call is handled as you intended.

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

SPA9000 supports call forward by client station:
•

unconditionally

•

when busy

•

on no answer

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

Call Transfer Support on SPA9000
•

”Call Forward Bridge Mode,” on page 72

•

”Call Transfer Bridge Mode,” on page 72

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 if it is between two different line
interfaces

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Managing Inbound Calls with Hunt Groups

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.
You can configure two types of Hunt Groups:
•

”Hunt Group that Rings All Stations Simultaneously,” on page 73

•

”Hunt Group that Rings Stations Individually,” on page 75

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 77.

Hunt Group that Rings All Stations Simultaneously
You can create a hunt group that simultaneously rings all stations in the group. Optionally, you
can assign a name to the group or use wildcard characters to create the list of stations in the
group.
Syntax for a “Ring All Stations” Hunt Group Rule
The basic syntax for this type of group is shown below. Optionally, you can assign a name to the
group.
SYNTAX: ext-in:name=groupname,hunt-list
•

ext-in (required): The extension number that users dial to contact this group

•

name (optional): The name of the group, which appears in the Corporate Directory on
the phones

•

hunt-list (required): The station numbers that participate in this group, separated by
commas

NOTE: To create multiple groups, separate the hunt group rules by typing the pipe
character: |
Examples of “Ring All Stations” Hunt Group Rules
•

Contacting a group of stations simultaneously
EXAMPLE: 500:101,102,103
This hunt group is assigned a number (500:). The group includes three stations
(101,102,103). A call to extension 500 rings all three stations simultaneously.

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•

Creating a named hunt group
EXAMPLE: 500:name=TechSupport,101,102,103
This hunt group is assigned a number (500:) and a name (name=TechSupport). The group
includes three stations (101,102,103). A call to extension 500 rings all three stations
simultaneously.

•

Wildcards in Hunt List
EXAMPLE 1: 500:10?
In this example, the ? symbol is a wildcard character that represent any single digit. A
call to extension 500 simultaneously rings all stations 100 through 109.
EXAMPLE 2: 500:1*
In this example, the * symbol is a wildcard character that represents any number of
digits. A call to extension 500 simultaneously rings all stations that begins with the
number 1.
EXAMPLE: 500:name=TechSupport,1*
In this example, the group has a name, Tech Support. The * symbol is a wildcard
character that represents any number of digits. A call to extension 500 simultaneously
rings all stations that begins with the number 1.

•

Creating multiple hunt groups
EXAMPLE: 500:name=TechSupport,101,102,103|600:name=AccountSupport,2*|
Two hunt groups are created and are assigned names. A call to extension 500
simultaneously rings stations 101, 102, and 103. A call to extension 600 simultaneously
rings all stations that begin with the number 2.

Creating a Hunt Group to Ring All Stations
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 24.)
2. Click Voice tab > SIP.
3. Scroll down to the PBX Parameters section.
4. In the Hunt Groups field, type your hunt group rule. See “Syntax for a “Ring All Stations” Hunt
Group Rule” on page_61 and “Examples of “Ring All Stations” Hunt Group Rules” on
page_61.
5. Click Submit All Changes. The phones reboot.
6. To verify the settings, place a call to the new group line. Verify that all of the listed
extensions ring simultaneously.

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Hunt Group that Rings Stations Individually
You can create a hunt group that rings the stations individually, in sequential order or random
order. You need to set the ring interval and the maximum number of cycles or seconds to occur
before a call is transferred to voice mail.
Syntax for a “Ring Individual Stations” Hunt Group Rule
The syntax for a “ring individual stations” hunt group rule is shown below. Optionally, you can
assign a name to the group.
SYNTAX: ext-in:name=groupname,hunt-list,type=algo;ring-interval;max,cfwd=ext-fwd
•

ext-in (required): The extension number that users dial to contact this group

•

name (optional): The name of the group, which appears in the Corporate Directory on
the phones

•

hunt-list (required): The station numbers that participate in this group, separated by

commas
•

type (required to ring individual stations): The type of routing among the stations in the

group
–

re (sequentially from the beginning): Every incoming call is routed to the first station

in the list. If the call is unanswered, it cycles through the stations in the listed order.
The syntax is hunt=re.
–

ne (sequentially from the next station): An incoming call is routed by determining

which station most recently answered a call for this group. The new call is routed to
the station that is next in the list. If the call is unanswered, it cycles through the
stations in the listed order. The syntax is hunt=ne.
–
•

ra (random): An incoming call is routed randomly to any listed station. If the call is
unanswered, it cycles through the stations in random order. The syntax is hunt=ra.

ring-interval (required to ring individual stations): The number of seconds that elapse

before an unanswered call is routed to another station in the group
•

max (required to ring individual stations): Either the maximum number of cycles or the

total number of seconds that elapse before an unanswered call is forwarded to the cfwd
destination. If the max value is less than the ring interval, it is interpreted as the number
of cycles. If the max value is greater than the ring interval, it is interpreted as the total
number of seconds.
•

cfwd (required to ring individual stations): The Call Forward destination for an

unanswered call. The destination can be another station, a voice mailbox, or the Auto
Attendant.
NOTE: Wildcard characters can be used in the hunt list. Multiple groups can be entered,
using the pipe character as a separator.

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Examples of “Ring Individual Stations” Hunt Group Rules
•

“From the Beginning” Order with Forwarding to the Auto Attendant
300:name=TechSupport,101,102,100,hunt=re;20;2,cfwd=aa

This hunt group is assigned an extension number (300:) and a name
(name=TechSupport,). The group includes three stations (101,102,100,). Calls are routed
sequentially from the beginning of the list (hunt=re;). Every new call is routed to 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 elapse, an unanswered call is forwarded
to the Auto Attendant (, cfwd=aa).
•

Random Order with Forwarding to a Voice Mailbox
400:name=Sales,101,102,103,hunt=ra;30;1,cfwd=vm2100

This hunt group is assigned an extension number (400:) and a name (name=Sales,). The
group includes three stations (101,102,103,). Calls are 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).
•

“Next Station” Order with Forwarding to a Station
500:name=Scheduling,102,103,101,hunt=ne;45;240,cfwd=100

This hunt group is assigned an extension number (500:) and a name (name=Scheduling,).
The group includes three stations (102,103,101,). Calls are routed in “next station” order
(hunt=ne;). Suppose that station 103 answered the most recent call. 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).
•

Multiple Lines with Wildcard Characters
400:name=CustomerService,10?,hunt=re;20;2,cfwd=aa|500:name=Sales,20?,hunt=ra;30;1,
cfwd=vm100| 600:name=TechSupport,300,301,302,hunt=ne;45;3,cfwd=100|

Three hunt groups are created. A call to extension 400 rings stations 100 to 109 in “from
the beginning order.” A call to extension 500 rings stations 200 to 209 in random order.
A call to extension 600 rings the listed stations in “next station” order.
Creating a Hunt Group to Ring Individual Stations
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 24.)
2. Click Voice tab > SIP.
3. Scroll down to the PBX Parameters section.

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Managing Inbound Calls with Shared Line Appearances

4. In the Hunt Groups field, enter the hunt group rule. See “Syntax for a “Ring Individual
Stations” Hunt Group Rule” on page_62 and “Examples of “Ring Individual Stations” Hunt
Group Rules” on page_63.
5. Click Submit All Changes. The SPA9000 and the phones reboot.
6. To verify the settings, place a call to the extension that you created. Verify that member
stations ring in the desired order, with the desired ring interval, maximum cycles, and call
forwarding.

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.
NOTE:
•

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 73.

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Managing Inbound Calls with Shared Line Appearances

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 24.)
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.
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, Linksys 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.
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

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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

NOTE: You can configure additional line keys for the same SLA.
5. Click Submit All Changes.
6. Repeat this procedure for each phone that you want to configure with the SLA.

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5

Connecting to the SPA400 Administration Web Server

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
LVS users the ability to receive and playback voice mail messages.
•

”Connecting to the SPA400 Administration Web Server,” on page 80

•

”Configuring the SPA400 Network Connection,” on page 81

•

”Managing Access to the SPA400 Web-Based Configuration Utility,” on page 83

•

”Upgrading the Firmware for the SPA400,” on page 84

•

”Configuring a SPA400 to Interoperate with the SPA9000,” on page 85

•

”Configuring a SPA400 for Voice Mail Service,” on page 87

•

”Managing the Voice Mail Messages on the USB Key,” on page 94

•

”Enabling Debugging on the SPA400,” on page 95

SEE ALSO: Appendix C, "SPA400 Field Reference"

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.

1. Start Internet Explorer, and enter the IP address of the SPA400 that you want to configure.
NOTE: Use Internet Explorer. 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.
2. When the password prompt appears, enter the default user name, Admin, with no
password. Then click OK.

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Configuring the SPA400 Network Connection

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 page 83.

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.
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.
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
managing system access, refer to the LVS Administration Guide.
3. Click Setup tab > Basic Setup.
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.

Domain Name Server (DNS) Address section:
•

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

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Saving or Discarding Changes on the SPA400

•

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

NTP section:
•

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.

5. 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.
•

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.

TIP: 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.
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.

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Managing Access to the SPA400 Web-Based Configuration Utility

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.
1. Connect to the SPA400 administration web server. (See ”Connecting to the SPA400
Administration Web Server,” on page 80.)
2. Click Administration tab > Management.
SPA400 Administration > Management

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.

4. Click Save Settings.
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.
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).

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Upgrading the Firmware for the SPA400

Upgrading the Firmware for the SPA400
As needed, you can download new firmware from Linksys.com and then install the firmware on
the SPA400.
1. Connect to the SPA400 administration web server. (See ”Connecting to the SPA400
Administration Web Server,” on page 80.)
2. Click Administration tab > Firmware Upgrade.
3. Click Browse.
SPA400 Administration > Firmware Upgrade

4. 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.

5. Click Upgrade.
6. When the confirmation message appears, click OK.

7. 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.

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Configuring a SPA400 to Interoperate with the SPA9000

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.
NOTES:
•

Complete this procedure for each SPA400, whether it is used as a PSTN gateway or as a
voice mail server.

•

You also must configure a SPA9000 line interface for this device. See ”Configuring a Line
Interface for a SPA400 (PSTN or Voice Mail),” on page 48.

•

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.

1. Connect to the SPA400 administration web server. (See ”Connecting to the SPA400
Administration Web Server,” on page 80.)
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 page 83.
3. Click Setup tab > SPA9000 Interface.
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
NOTES:
–

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.

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–

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.

–

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

5. Click Save Settings at the bottom of the page.
6. Click Setup tab > Voice.
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

8. Click Save Settings at the bottom of the page.
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.
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.
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 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.

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Configuring a SPA400 for PSTN Access

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.

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 page 87.

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.

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.

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 48.

This section includes the following topics:
•

”Voice Mail Capacity,” on page 88

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Configuring a SPA400 for Voice Mail Service

•

”Configuring Local Voice Mail Service on a SPA400,” on page 88

•

”Setting Up Voice Mail on Each Station,” on page 90

•

”Enabling Remote Voice Mail Access (Optional),” on page 93

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 60second 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.

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 48.
1. Connect to the SPA400 administration web server. (See ”Connecting to the SPA400
Administration Web Server,” on page 80.)
2. Click Setup tab > Voicemail Server.
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.

•

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.

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Configuring a SPA400 for Voice Mail Service

SPA400 Setup > Voicemail Server: Voicemail Setting

NOTE: These settings correspond to the settings on the SPA9000 Voice > Line page.
SPA400 > Setup > Voicemail Server

SPA9000 > Voice > Line N

4. Click Save Settings.
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

NOTE: Later you configure each client station for voice mail access to the mail boxes
that you create on this page.
6. Click Save Settings at the bottom of the page.
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Configuring a SPA400 for Voice Mail Service

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.
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.
9. Continue to the next procedure, ”Setting Up Voice Mail on Each Station,” on page 90.

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.
1. Connect to the SPA9000 administration web server. (See ”Connecting to the SPA9000
Administration Web Server,” on page 24.)
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.

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Configuring a SPA400 for Voice Mail Service

SPA9000 PBX Status

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

4. 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.

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Configuring a SPA400 for Voice Mail Service

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.

Phone Configuration > Ext N
SPA400 > Setup > Voicemail Users

6. Click Submit All Changes. The phone reboots.
7. Close the browser window for this station.
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.
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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Configuring a SPA400 for Voice Mail Service

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 the Voice Mail Server and Voice Mail Users on the SPA400” on page_53)
and the Phone configuration page (see “Setting Up Each Station” on page_55).

–

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).

1. Connect to the SPA9000 administration web server. (See ”Connecting to the SPA9000
Administration Web Server,” on page 24.)
2. Click Voice tab > SIP.
3. Scroll down to the Auto Attendant Parameters section.
4. Edit the AA Dial Plan 1 string to include a code for the voice mail server, as described below.
•

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.

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Managing the Voice Mail Messages on the USB Key

•

MailboxManageNumber: The Mailbox Manage Number that was entered on the SPA400
Voice mail Settings page. If you followed the instructions in the procedure “Configuring
the SPA400” on page_48, 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.

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 page 117.
6. Click Voice tab > Line N, where N is the number of the line interface for this SPA400 unit.
7. Make sure that the VMSP Bridge, XFER Bridge Mode, and CFWD Bridge Mode fields are set to
all.
8. Click Submit All Changes. The SPA9000 and the phones reboot.
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.

Managing the Voice Mail Messages on the USB Key
To delete unneeded voice mail messages on the USB key, refer to the procedure below.
1. Connect to the SPA400 administration web server. (See ”Connecting to the SPA400
Administration Web Server,” on page 80.)
2. Click Administration tab > USB Setting.

3. Check the I want to reset USB check box.
4. Click Reset.

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Enabling Debugging on the SPA400

Note

All the voicemail of all users will be deleted. The deleted voicemail
messages are not recoverable.

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

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.

•

If you are interested in working with these logs yourself, you will need to use an
application that reads syslogs. Linksys partners can download a syslog server tool
(slogsrv.exe) from the Linksys Partner Connection. (VARs, see Product Utilities > Linksys
Voice System > SPA Utilities > Syslog Server for SPA Devices. SPs, see Technical Tools >
SPA Utilities > Syslog Server for SPA Devices.)

Note

As a best practice, enable logging only when needed, and disable
logging when you finish the investigation. Logging information can
impact system performance.

1. Connect to the SPA400 administration web server. (See ”Connecting to the SPA400
Administration Web Server,” on page 80.)
2. Click Event Logs tab.
3. Use the drop-down lists to choose the types of information that you want to collect.

4. Click Save Settings.
5. Click Setup tab > Basic Setup.
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Enabling Debugging on the SPA400

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.

7. Start a command prompt on the PC, and run the system logging software.
8. Run the scenario that causes the problem and when done, press CTRL+C in the command
window to stop the logs.
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

6

Using the Internal Music Source for Music On Hold

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 page 97

•

”Configuring a Streaming Audio Server,” on page 99

Using the Internal Music Source for Music On Hold
The SPA9000 has an internal music source with the user ID imusic that plays an internally
stored music file repeatedly. The SPA9000 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 page 97

•

”Changing the Music File for the Internal Music Source,” on page 98

•

”Restoring the Original Music File,” on page 98

Using the Internal Music Source
The SPA9000 has an internal music source that plays an internally stored music file repeatedly.
The SPA9000 ships with a default music file (Romance de Amor).
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.
2. Start Internet Explorer, and then enter the IP address of the telephone. The Telephone
Configuration page appears in a separate browser window.
3. Click Admin Login, and then click Advanced.
4. Click the Ext 1 tab.
5. Scroll down to the Call Feature Settings section.
6. Enter the following value in the MOH Server field: imusic
7. Click Submit All Changes.
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).

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Using the Internal Music Source for Music On Hold

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 with no file header, up to
65.5 seconds in length, with no header information

1. Before you begin, make sure that you have TFTP server software running on your computer.
2. Connect to the SPA9000 administration web server, and choose Admin access with
Advanced settings. (See ”Connecting to the SPA9000 Administration Web Server,” on
page 24).
3. Click Voice tab > SIP.
4. Scroll down to the Internal Music Source Parameters section.
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)

•

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

6. Click Submit All Changes. The SPA9000 reboots. Then the SPA9000 downloads the file and
stores it in flash memory.

Restoring the Original Music File
If you want to restore the original default music file for the internal music source, you can
download the file from Linksys.com.
1. On Linksys.com, choose Support > Technical Support.
2. Click Choose A Product.
3. Choose IP PBX from the Voice over IP drop-down list.
4. Under Downloads, choose the Click Here link.
5. Choose the correct version in the drop-down list.
6. Under Others, click the Default Music File link.

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Configuring a Streaming Audio Server

7. Unzip this file and save it to an accessible directory on your TFTP server. Then repeat this
procedure and enter the correct server IP address, port, and the path to the unzipped file.

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

”Using a Streaming Audio Server,” on page 99

•

”Using the IVR with an SAS Line,” on page 100

•

”Example SAS with MOH,” on page 100

•

”Configuring the Streaming Audio Server,” on page 101

Using a Streaming Audio Server
On a connected call, the SPA9000 may place the remote party on hold by performing a hookflash to initiate a three-way call or by swapping two calls during call-waiting. If the remote
client indicates that it can still receive audio while the call is holding, the SPA9000 can be
configured to contact an auto-answering streaming audio server (SAS) to stream audio to the
holding party. When used this way, the SAS is referred to as an MOH Server.
The SAS feature lets you use attach an audio source to one of the SPA9000 FXS ports (Phone 1
or Phone 2 on the PAP2T) and use it as a streaming audio source device. If the SPA9000 has
multiple FXS ports, either or both of the associated lines (Line 1 and Line 2 on the PAP2T) can
be configured as an SAS server.
When the line is called and the FXS port is off hook, the SPA9000 answers the call automatically
and streams audio to the calling party.
If the FXS port is on-hook when the incoming call arrives, the SPA9000 replies with a SIP 503
response code to indicate “Service Not Available.” If an incoming call is auto-answered, but later
the FXS port changes to on-hook, the SPA does not terminate the call but continues to stream
silence packets to the caller. If an incoming call arrives when the SAS line has reached full
capacity, the SPA replies with a SIP 486 response code to indicate “Busy Here.”
The SAS line can be setup to refresh each streaming audio session periodically using a SIP reINVITE 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.
Each SAS server can maintain up to five simultaneous calls. If the second line on the SPA9000 is
disabled, then the SAS line can maintain up to 10 simultaneous calls. Further incoming calls
receive a busy signal (SIP 486 Response).
The streaming audio source must be off-hook for the streaming to occur. Otherwise incoming
calls get a error response (SIP 503 Response). The SAS line does not ring for incoming calls even
if the attached equipment is on-hook.

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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.
Set up the Proxy and Subscriber Information for the SAS Line as you normally would with a
regular user account.
Call Forwarding, Call Screening, Call Blocking, DND, and Caller-ID Delivery features are not
available on an SAS line.

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:
1. Power off the SPA9000.
2. Connect a phone to the port and make sure the phone is on-hook.
3. Power on the SPA9000.
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.

Example SAS with MOH
Figure 6-1 MOH Application with a SPA9000 Line Configured as an SAS
SPA2:
IP=192.168.2.200
UserID[1]=2001, SIP Port[1]=5060
UserID[1]=2002, SIP Port[1]=5061

SPA1:
IP=192.168.2.100
UserID[1]=1001, SIP Port[1]=5060
UserID[1]=1002, SIP Port[1]=5061

IP Network
Phone 1

Phone 1
Phone 2

Phone 2

Media
signal
adapter
Line in
Music
source

In this example, the SAS Line is registered with the SPA9000 as the other subscribers.

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On SPA1:
SAS Enable[1] = no
MOH Server [1] = 1002
SAS Enable[2] = yes

On SPA2:
SAS Enable[1] = no
MOH Server [1] = 1002
SAS Enable[2] = no
MOH Server [2] = 1002

Configuring the Streaming Audio Server
The following provides step-by-step procedures for implementing a SAS with an external music
source.
1. Connect an RJ-11 adapter between the music source (a CD player or iPod, for example) and
an FXS port on the SPA9000 (Phone 1 or Phone 2).
2. Connect to the SPA9000 administration web server, and choose Admin access with
Advanced settings. (See ”Connecting to the SPA9000 Administration Web Server,” on
page 24).
3. Configure the FXS port:
a. Click Voice tab > FXS N, where N represents the SPA9000 phone port number where
you connected the RJ-11 adapter.
b. 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 other
extension.

c. Scroll down to the Streaming Audio Server (SAS) section.
d. From the SAS Enable drop-down list, choose yes.
Voice > FXS N > Streaming Audio Server

4. Click Submit All Changes.

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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.

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

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Configuring the LVS Auto-Attendant

7

How the Auto Attendant Works

Configuring the LVS Auto-Attendant
This chapter describes how to configure the LVS Auto Attendant using the IVR, and XML
scripting.
This chapter contains the following sections:
•

”How the Auto Attendant Works,” on page 103

•

”Working with the Auto Attendant Greetings,” on page 104

•

”Writing an Auto Attendant Script,” on page 108

•

”Configuring the DayTime, NightTime and Weekend/Holiday Auto Attendants,” on
page 116

•

”Configuring Dial Plans for the Auto Attendant,” on page 117

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 Auto-Attendants.
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.

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 66.

•

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 page 104.

•

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

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page, Auto Attendant Parameters section, AA Script 1 - 3 fields. For more information, see
”Writing an Auto Attendant Script,” on page 108.
•

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
page 116.

•

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 page 117.

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 page 104

•

”Recording an Auto Attendant Prompt,” on page 105

•

”Downloading Prompts,” on page 107

Using Pre-Recorded Prompts
You can save up to ten Auto Attendant prompts. Four pre-recorded prompts are provided, as
listed in Table 7-1.

Table 7-1 Prompts
Prompts ID

Default Audio Content

Prompt1
Prompt2
Prompt3
Prompt4

“If you know your party’s extension, you may enter it now.”
“Your call has been forwarded.”
“Not a valid extension, please try again.”
“Goodbye.”

Note

These prompts are available in languages other than English. See
”Localizing the SPA9000 Auto Attendant Prompts,” on page 118.

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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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.

1. Connect an analog phone to the Phone 1 or Phone 2 port of the SPA9000.
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.
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.
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.
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 #.
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 re-record, enter 3. To
exit, enter *.
7. Press 1 to save the new recorded message.
The IVR plays the following prompt: One moment, please.
8. Wait for several seconds while the save is completed.

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Example Prompts
In this example, the business will have different Auto Attendant Scripts for different times of
day.

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 page 109.

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Using the IVR Prompts to Change Recordings
The IVR prompts guides you through the process of recording, erasing, and changing greetings.
Refer to Figure 7-1.
Figure 7-1 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

User enters 3

Menu-3-A
Play prompt (“You may record your
message after the tone. When
finished, press #.”)

Retrieve the
prompt data and
play prompt

User enters 2

Play prompt
(“Value saved”)

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

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.

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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 24).
2. Click Voice tab > SIP.
3. Scroll down to the Auto Attendant Parameters section.
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.
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 page 109

•

”Elements of XML Scripting Grammar ,” on page 111

•

”Auto Attendant XML Instructions Set,” on page 113

•

”Entering an Auto Attendant Script,” on page 115

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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.
SEE ALSO: ”Elements of XML Scripting Grammar ,” on page 111
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.

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.


Linksys Voice System Administration Guide

extension, please try again.

The form is ended.
The script is ended.

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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.

Script Elements

Purpose



This script is for the AA.



This form is given the name “DIR” for Directory.
It is a menu type of form, which accepts DTMF

inputs.
”

Not a valid extension, please try again.

•
•

If the user presses *, the Auto Attendant
plays Prompt 4: Good-bye.
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 page 111

•

”Action Instruction,” on page 112

•

”Noinput Instruction ,” on page 112

•

”Nomatch Instruction,” on page 113

•

”Menu Matched Instruction for Touch Tone (DMTP) Input,” on page 113

Audio Instruction
The following is an example of the audio instruction:

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