Copyright and Trademarks

Specifications are subject to change without notice. Linksys is a registered trademark or trademark of Cisco Systems, Inc. and/or its affiliates in the U.S. and certain other countries. Copyright © 2006 Cisco Systems, Inc. All rights reserved. Other brands and product names are trademarks or registered trademarks of their respective holders.

WARNING: This product contains chemicals, including lead, known to the State of California to cause cancer, and birth defects or other reproductive harm. Wash hands after handling.

How to Use this Guide

The guide to the IP Telephony System has been designed to make understanding networking with the IP Telephony System easier than ever. Look for the following items when reading this User Guide:

This checkmark means there is a note of interest and is something you should pay special attention to while using the IP Telephony System.

This exclamation point means there is a caution or warning and is something that could damage your property or the IP Telephony System.

This question mark provides you with a reminder about something you might need to do while using the IP Telephony System. In addition to these symbols, there are definitions for technical terms that are presented like this:

word: definition.

Also, each figure (diagram, screenshot, or other image) is provided with a figure number and description, like this:

Figure 0-1: Sample Figure Description

Figure numbers and descriptions can also be found in the “List of Figures” section.

SPA9000-UG-60303B JL

Table of Contents

: Introduction1
Welcome 1
What’s in this Guide? 2
: Applications for the IP Telephony System4
How Does the IP Telephony System Fit into My Business or Home? 4
What Does the IP Telephony System Do? 4
A Typical Scenario 4
Which Call Management Features Does the IP Telephony System Offer? 5
: Getting to Know the IP Telephony System6
The Back Panel 6
The Front Panel 7
: Getting Started8
Overview 8
Before You Begin 8
Instructions for Installing the IP Telephony System 9
Receiving and Handling External Phone Calls 13
Configuring the Auto-Attendant 13
: Using the Interactive Voice Response Menu14
Overview 14
Accessing the Interactive Voice Response Menu 14
Using the Interactive Voice Response Menu 14
Entering a Password 19
Configuring the Settings for Your Internet Phone Service 19
Configuring the Auto-Attendant Messages 20
: Using the Web-based Utility22
Overview 22
How to Access the Web-based Utility 23
The PBX Status Screen 23
The Router Tab 25
The Voice Tab 31
Appendix A: Troubleshooting 70
Common Problems and Solutions 70
Frequently Asked Questions 80
Appendix B: Configuring the Nighttime Auto-Attendant 83
Description of the Auto-Attendant 83
Instructions for Setting Up the Nighttime Auto-Attendant 83
Appendix C: Dial Plan and Auto-Attendant Scripting for Advanced Users 87
Overview 87
Configuring Dial Plans 87
Configuring Dial Plans for the Auto-Attendant 89
Configuring the Auto-Attendant 89
Appendix D: New Music for the Music-on-Hold Feature 99
Overview 99
Before You Begin 99
Instructions for Converting the Music File 100
Instructions for Configuring the IP Telephony System 100
Appendix E: Finding the MAC Address and IP Address for Your Ethernet Adapter 102
Windows 98 or Me Instructions 102
Windows 2000 or XP Instructions 102
For the Router’s Web-based Utility 103
Appendix F: Windows Help 104
Appendix G: Glossary 105
Appendix H: Specifications 110
Appendix I: Warranty Information 114
Appendix J: Regulatory Information 115
Appendix K: Contact Information 121
Internet Telephony Service Provider (ITSP) 121
Linksys 121

List of Figures

Figure 2-1: A Scenario for the IP Telephony System 4 Figure 3-1: Back Panel 6 Figure 3-2: Front Panel 7 Figure 4-1: A Typical Scenario for the IP Telephony System 8 Figure 4-2: Connect to the Phone 1 Port 9 Figure 4-3: Connect to the Internet Port 9 Figure 4-4: Connect to the Ethernet Port 9 Figure 4-5: Connect to Power 9 Figure 4-6: Voice - SIP Screen - PBX Parameters 10 Figure 4-7: Router - WAN Setup Screen 10 Figure 4-8: Voice - Line 1 Screen 12 Figure 5-1: Auto-Attendant Options 18 Figure 5-2: Auto-Attendant Message Options 21 Figure 6-1: PBX Screen - Parking Lot 24 Figure 6-2: PBX Screen - Inbound Call 24 Figure 6-3: PBX Screen - Outbound Call 24 Figure 6-4: Router - Status Screen 25 Figure 6-5: Router - WAN Setup Screen 27 Figure 6-6: Router - LAN Setup Screen 29 Figure 6-7: Router - Application Screen 30 Figure 6-8: Voice - Info Screen - Product Information 31 Figure 6-9: Voice - Info Screen - System Status 31 Figure 6-10: Voice - Info Screen - FXS Status 32 Figure 6-11: Voice - Info Screen - Line Status 33 Figure 6-12: Voice - Info Screen - Auto Attendant Prompt Status 33 Figure 6-13: Voice - System Screen 34 Figure 6-14: Voice - SIP Screen - SIP Parameters 35 Figure 6-15: Voice - SIP Screen - SIP Timer Values 36

Figure 6-16: Voice - SIP Screen - Response Status Code Handling 37 Figure 6-17: Voice - SIP Screen - RTP Parameters 37 Figure 6-18: Voice - SIP Screen - SDP Payload Types 37 Figure 6-19: Voice - SIP Screen - NAT Support Parameters 38 Figure 6-20: Voice - SIP Screen - PBX Parameters 39 Figure 6-21: Voice - SIP Screen - Auto Attendant Parameters 42 Figure 6-22: Voice - SIP Screen - PBX Phone Parameters 44 Figure 6-23: Voice - Provisioning Screen - Configuration Profile 45 Figure 6-24: Voice - Provisioning Screen - Firmware Upgrade 46 Figure 6-25: Voice - Provisioning Screen - General Purpose Parameters 47 Figure 6-26: Voice - Regional Screen - Call Progress Tones 48 Figure 6-27: Voice - Regional Screen - Distinctive Ring Patterns 49 Figure 6-28: Voice - Regional Screen - Distinctive Call Waiting Tone Patterns 50 Figure 6-29: Voice - Regional Screen - Distinctive Ring/CWT Pattern Names 50 Figure 6-30: Voice - Regional Screen - Ring and Call Waiting Tone Spec 51 Figure 6-31: Voice - Regional Screen - Control Timer Values 51 Figure 6-32: Voice - Regional Screen - Vertical Service Activation Codes 52 Figure 6-33: Voice - Regional Screen - Vertical Service Announcement Codes 55 Figure 6-34: Voice - Regional Screen - Outbound Call Codec Selection Codes 55 Figure 6-35: Voice - Regional Screen - Miscellaneous 56 Figure 6-36: Voice - FXS 1 Screen - Network Settings 59 Figure 6-37: Voice - FXS 1 Screen - SIP Settings 59 Figure 6-38: Voice - FXS 1 Screen - Subscriber Information 60 Figure 6-39: Voice - FXS 1 Screen - Dial Plan 61 Figure 6-40: Voice - FXS 1 Screen - Streaming Audio Server 61 Figure 6-41: Voice - FXS 1 Screen - Call Feature Settings 61 Figure 6-42: Voice - FXS 1 Screen - Audio Configuration 62 Figure 6-43: Voice - FXS 1 Screen - FXS Port Polarity Configuration 64 Figure 6-44: Voice - Line 1 Screen - Network Settings 65 Figure 6-45: Voice - Line 1 Screen - SIP Settings 65

Figure 6-46: Voice - Line 1 Screen - Subscriber Information 66 Figure 6-47: Voice - Line 1 Screen - Dial Plan 68 Figure 6-48: Voice - Line 1 Screen - NAT Settings 68 Figure 6-49: Voice - Line 1 Screen - Proxy and Registration 68 Figure B-1: Auto-Attendant Message Options 84 Figure B-2: Voice - SIP Screen - Auto Attendant Parameters 85 Figure E-1: IP Configuration Screen 102 Figure E-2: MAC Address/Adapter Address 102 Figure E-3: MAC Address/Physical Address 103 Figure E-4: MAC Address Clone 103

Chapter 1: Introduction

Welcome

Thank you for choosing the Linksys IP Telephony System. The System combines the rich feature set of legacy PBX (Private Branch eXchange) telephone systems with the convenience and cost advantages of Internet telephony. It supports common key system features such as an auto-attendant, music-on-hold, call forwarding, three-way call conferencing, and more.

The System is so easy to configure that a fully working system can be set up in minutes. New Linksys SPA-family Internet telephones are automatically detected and registered when they are connected to the System. While the System will work with any SIP-compatible Internet telephone, it is the ideal host for Linksys business telephones, including model number: SPA941. The System supports the advanced features of these phones, such as shared line appearances, hunt groups, call transfer, call park, and group paging. Plus, with its two FXS ports, the System can support traditional analog devices such as telephones, fax machines, answering machines, media adapters.

How does the System do all of this? By connecting your analog phones or fax machines to the System and connecting the System and Internet phones to your router, then the System can direct voice communications for your network.

But what does all of this mean?

Networks are useful tools for sharing Internet access and computer resources. Multiple computers can share Internet access, so you don’t need more than one high-speed Internet connection. With Internet phone service, your Internet access can now be shared by your Internet phones as well. You will be able to make phone calls using your Internet phone service account, even while another colleague is web browsing. Plus, you can access one printer from different computers and access data located on another computer’s hard drive (with the right permissions).

PCs on a wired network create a LAN, or Local Area Network. They are connected with Ethernet cables, which is why the network is called “wired”. The System takes your wired network and lets you integrate Internet phones and Internet phone service.

When you first install the System, Linksys strongly recommends that you use the Setup Wizard, which you can download from www.linksys.com. If you do not wish to run the Setup Wizard, then use the instructions in the Quick Installation or this User Guide to help you. These instructions should be all you need to get the most out of the IP Telephony System.

NOTE: Some of these features are set up from the Internet phones.

network: a series of computers or devices connected for the purpose of data sharing, storage, and/or transmission between users.

lan (local area network): the computers and networking products that make up the network in your home or office.

ethernet: an IEEE standard network protocol that specifies how data is placed on and retrieved from a common transmission medium.

What’s in this Guide?

This user guide covers the steps for setting up a network with the System. Most users will only need to use “Chapter 4: Getting Started.” When you’re finished, then you are ready to make calls within your system as well as calls to the outside world.

You also have other chapter available for reference:

  • Chapter 1: Introduction This chapter describes the System and this User Guide.
  • Chapter 2: Applications This chapter discusses the most common scenarios for the System.
  • Chapter 3: Getting to Know the IP Telephony System This chapter describes the physical features of the System.
  • Chapter 5: Using the Interactive Voice Response Menu This chapter explains how to configure the System’s network settings when you access its Interactive Voice Response Menu.
  • Chapter 6: Using the Web-based Utility This chapter explains how to configure the settings of the System through the Web-based Utility.
  • Appendix A: Troubleshooting This appendix describes some possible problems and solutions, as well as frequently asked questions, regarding installation and use of the System.
  • Appendix B: Configuring the Nighttime Auto-Attendant This appendix explains how to set up the auto-attendant for nighttime (non-business) hours.
  • Appendix C: Dial Plan and Auto-Attendant Scripting for Advanced Users This appendix explains how to define the dial plan and auto-attendant instructions. (These instructions are for advanced users only.)
  • Appendix D: New Music for the Music-on-Hold Feature This appendix explains how to replace the System’s default music file with your own music file.
  • Appendix E: Finding the MAC Address and IP Address for Your Ethernet Adapter This appendix instructs you on how to find the MAC address or Ethernet address of your PC’s Ethernet network adapter.
  • Appendix F: Windows Help This appendix describes how you can use Windows Help for instructions about networking, such as installing the TCP/IP protocol.
  • Appendix G: Glossary This appendix gives a brief glossary of terms frequently used in networking.
  • Appendix H: Specifications This appendix provides the technical specifications for the System.
  • Appendix I: Warranty Information This appendix supplies the warranty information for the System.
  • Appendix J: Regulatory Information This appendix supplies the regulatory information regarding the System.
  • Appendix K: Contact Information This appendix provides contact information for a variety of Linksys resources, including Technical Support.

Chapter 2: Applications for the IP Telephony System

How Does the IP Telephony System Fit into My Business or Home?

High-speed Internet access is a valuable resource. When you have more than one computer, chances are you want to share that Internet access with all of your computers. That’s when you create a network, a collection of devices connected to each other. A device called a router connects computers and other devices, so they can Internet share a high-speed Internet connection and other resources, including data and printers.

One of the biggest benefits of the Internet is data communications, either e-mail or web browsing, whether you send a file to a client or download the latest software upgrade. With the System, you also get voice communications.

SPA941 Cable/DSL Modem

What Does the IP Telephony System Do?

The System connects multiple Internet phones to an Internet phone service. The System manages and routes all calls. Incoming calls go to the auto-attendant, an automated greeting system, or correct internal extension (each

SPA941 Switch Router Desktop phone has its own extension number). Outgoing calls go to the correct external phone number (you can have more than one external phone number). Computer

You can have not only more than one external phone number, but also up to four Internet Telephony Service Providers (ITSPs) for maximum flexibility.

NOTE: The basic configuration of the System lets you connect up to four Internet phones and use SPA941 up to four ITSPs. To expand the basic configuration, contact your primary ITSP for more information.

A Typical Scenario

Typically, you connect the Internet port of the System to a local network port of your router. Then connect a switch to another local network port of your router. Use this switch to connect Internet phones, computers, and other devices. Then connect an administration computer to the Ethernet port of the System.

If you have analog telephones or fax machines, you can connect them to the Phone ports, so you can also use Analog Fax Administration those phones to make Internet phone or fax calls. (More details are available in “Chapter 4: Getting Started.”) Phone Computer

Figure 2-1: A Scenario for the IP Telephony System

What Kind of Router Should I Use?

For your network, get the highest-performance router possible. For best results, use a QoS (Quality of Service) router, so it can assign top priority to voice traffic.

What Kind of Switch Should I Use?

Again, performance is key. For best results, use a switch that offers QoS (Quality of Service) and full wire-speed switching. QoS enables the switch to give top priority to voice traffic, while full wire-speed switching lets it forward packets as fast as your network can deliver them. The next best choice is a switch featuring QoS (Quality of Service).

What if I Keep My Traditional Phone Service?

Traditional phone service, also known as Plain Old Telephone Service (POTS), runs on a network called the Public Switched Telephone Network (PSTN). If you decide to keep traditional phone service, then connect the Analog Telephone Adapter (model number: SPA3000) to the switch. (For more information, refer to the SPA3000 documentation.)

Which Call Management Features Does the IP Telephony System Offer?

Beyond basic call routing, the System offers several powerful and sophisticated features:

  • Auto-Attendant. An automated system guides each caller to the appropriate contact.
  • Music-on-Hold. You can combine the auto-attendant feature with the music- or information-on-hold feature, so the caller has a richer experience with your call system.
  • Call Hunt. You can designate which Internet phones receive outside calls. You can even have calls ring multiple phones, either simultaneously or one at a time.
  • Paging. When you want to page all of the Internet phones, you can use the System.
  • Dial Plans. When you have more than one dial plan, you can route outgoing calls to take advantage of the best rates available for the different types of calls.

After setup of the System, you will have dynamic and feature-rich Internet voice communications for your business or home.

NOTE: If your ITSP configured the System for you, then these features may already be set up. Check with your ITSP for more information.

(To set up these features yourself, refer to “Chapter 6: Using the Web-based Utility.”)

Chapter 3: Getting to Know the IP Telephony System

The Back Panel

The System’s ports are located on its back panel.

Figure 3-1: Back Panel
PHONE 1/2 The PHONE 1/2 ports allow you to connect analog telephones (or fax machines) to the System
using RJ-11 telephone cables (not included).
ETHERNET The ETHERNET port connects to an administration computer, so you can access the System’s
Web-based Utility for configuration.
INTERNET This INTERNET port connects to either a router or broadband modem.
Power The Power port is where you will connect the power adapter.

The Front Panel

The System’s LEDs are located on its front panel.

Figure 3-2: Front Panel
Power Green. The power LED is solidly lit when the System is powered on and connected to the
Internet. It flashes when there is no Internet connection.
ETHERNET Green. The ETHERNET LED is solidly lit when there is an Internet connection. It flashes when
there is network activity.
PHONE 1/2 Green. The PHONE 1/2 LED is solidly lit when the phone is on-hook and registered. (The
connection is registered if your Internet phone service account is active.) The LED is not lit
when the phone is on-hook and not registered. It flashes when the phone is off-hook.

Chapter 4: Getting Started

Overview

For first-time installation of the System, Linksys strongly recommends using the Setup Wizard, which you can download from www.linksys.com. For advanced users, you may follow the instructions in this chapter, and then Internet use the Web-based Utility for additional configuration (refer to “Chapter 6: Using the Web-based Utility”). To use the Interactive Voice Response Menu, proceed to “Chapter 5: Using the Interactive Voice Response Menu.”

Before You Begin

SPA941 Cable/DSL Modem Make sure you have the following:

  • IP Telephony System (model number: SPA9000)
    • One or more Internet phones (for example, Linksys SPA-family IP Phones, model number: SPA941)
    • • A router and cable/DSL modem (or gateway) SPA941 Switch Router Desktop Computer
    • One or more Ethernet network switches (so you can connect Internet phones or computers)
    • NOTE: For best results, use a switch that offers QoS (Quality of Service) and full wire-speed switching. QoS enables the switch to give top priority to voice traffic, while full wire-speed SPA941 switching lets it forward packets as fast as your network can deliver them. The next best choice is a QoS (Quality of Service) switch.
  • At least one active Internet phone service account and its settings if you want to make external calls
  • An active Internet connection if you want to make external calls
  • At least one computer for configuration of the System and Internet phones
  • Two or more Ethernet network cables
  • Analog telephones or fax machines (optional)

Analog Fax Administration Phone Computer

Instructions for Installing the IP Telephony System ip (internet protocol): a protocol used to send data over a network.

Internal Calls ip address: the address used to identify a computer or device on a network.

To install the System for internal calls, you will do the following:

  • connect and configure the System
  • connect the Internet phones Connect and Configure the System
  1. (optional) Plug an analog telephone into the Phone 1 port of the System.
    1. (optional) If you have a second analog telephone or fax machine, plug it into the Phone 2 port of the System.
    2. IMPORTANT: Do not connect the Phone port to a telephone wall jack. Make sure you only connect a telephone or fax machine to the Phone port. Otherwise, the System or the telephone wiring in your home or office may be damaged.
  2. Connect an Ethernet network cable to the Internet port of the System. Then connect the other end of the cable to one of the Ethernet ports on your router.
  3. Connect a different Ethernet network cable to the Ethernet port of the System. Then connect the other end to Figure 4-3: Connect to the Internet Port the computer you will use to manage the System (this will be called the administration computer).
  4. Connect the included power adapter to the Power port of the System, and then plug the power adapter into an electrical outlet.
  5. Launch the web browser on the administration computer.

7. Enter 192.168.0.1/admin/voice/advanced in the Address field (192.168.0.1 is the default local IP address of the System). Then press the Enter key. Figure 4-4: Connect to the Ethernet Port

  1. The Voice - Info screen will appear. Click the SIP tab.
  2. In the PBX Parameters section, select WAN from the Proxy Network Interface drop-down menu.

10. Click the Submit All Changes button.

  1. The Voice - Info screen will appear. Click the Router tab.
  2. Click the WAN Setup tab.

13. From the Connection Type drop-down menu, select Static IP. Figure 4-6: Voice - SIP Screen - PBX Parameters

14. In the Static IP Settings section, complete the Static IP, NetMask, and Gateway fields. Static IP. Enter a static IP address appropriate for your network. Write this down; you will use it later. NOTE: Make sure your router will not assign the System’s static IP address to any other

network device. For example, you can assign a static IP address outside of your router’s DHCP IP address range; however, it must be within the router’s subnet range. For more information about IP addressing, refer to the router’s documentation.

NetMask. Enter the subnet mask of your network router. Gateway. Enter the local IP address of your network router or gateway.

  1. In the Optional Settings section, complete the Primary DNS field. Primary DNS. Enter the DNS IP address of your network router.
  2. In the Remote Management section, select yes from the Enable WAN Web Server drop-down menu. Figure 4-7: Router - WAN Setup Screen

17. Click the Submit All Changes button.

18. The Router - Status screen will appear. Verify that the following settings match your entries:

  • WAN Connection Type - Static IP
  • Current IP
  • Current Netmask
  • Current Gateway
  • Primary DNS

Proceed to the next section, “Connect the Internet Phones.”

Connect the Internet Phones

  1. Connect an Ethernet network cable to one of the Ethernet ports on your router. Then connect the other end of the cable to an Ethernet port on a network switch.
  2. Connect the switch’s power adapter to its power port, and then plug the power adapter into an electrical outlet.
    1. Connect an Ethernet network cable to an Internet phone. Then connect the other end to one of the Ethernet ports on the switch.
    2. (If the Internet phone has been used before, reset it to its factory default settings first. Refer to its documentation for more information.)
  3. Connect the Internet phone’s power adapter to its power port, and then plug the power adapter into an electrical outlet.
  4. The Internet phone will reboot two to three times (each reboot may take up to one minute). The System will automatically assign an extension number to the Internet phone. When the Internet phone displays it extension number, then it is ready for use.

NOTE: The System automatically registers Linksys SPA-family Internet phones (including

NOTE: The default SIP port of the System

model number SPA941). If you connect a different SIP-compatible phone, then registration

is 6060.

will be manual. Refer to the documentation for your phone.

6. Repeat steps 3-5 until you have installed all of your Internet phones.

Congratulations! Now you can make calls from one Internet phone to another by dialing an extension number.

Continue to the next section to configure the System for external calls.

External Calls

For external calls, make sure you have an active Internet connection. Then configure the settings for your Internet phone service account on the System.

  1. Launch the web browser on the administration computer.
  2. Enter <IP address of the System>/admin/voice/advanced in the Address field (use the static IP address you previously assigned to the System). Then press the Enter key.
  3. The Voice - Info screen will appear. Click the Line 1 tab.
  4. On the Line 1 screen, enter the settings for your Internet phone service account. Subscriber Information

User ID. Enter the user ID (also called the account number) supplied by your ITSP. Do not use any hyphens, spaces, or other punctuation. Password. Enter the case-sensitive password supplied by your ITSP. Proxy and Registration

Proxy. Enter the proxy address supplied by your ITSP. If your ITSP supplied additional settings, enter those as well. Refer to the instructions your ITSP gave you.

  1. Click the Submit All Changes button to save your new settings.
  2. The System will reboot itself. Then the Internet phones will reboot themselves.
  3. The Voice - Info screen will appear. In the Line 1 Status section, make sure that the Registration Status says, “Registered.”

You are now ready to make your first external call. Use any phone connected to the System, and dial 9 first when you make an external call with the default US dial plan.

You can use analog telephones to make external calls; however, you cannot receive calls on any analog telephones unless you configure the appropriate settings. Refer to the Voice - FXS 1 section of “Chapter 6: Using the Web-based Utility” for instructions.

Congratulations! Now you can make external calls using the System.

NOTE: If your Internet Telephony Service Provider (ITSP) supplied the System, then it may be pre-configured for you, and you do not need to change any settings. Refer to the instructions supplied by your ITSP for more information.

NOTE: If you cannot make calls with the default US dial plan, visit www.linksys.com/kb for additional dial plans, or refer to “Appendix C: Dial Plan and Auto-Attendant Scripting for Advanced Users” to write your own script.

Receiving and Handling External Phone Calls

To receive external phone calls, you need to know the Direct Inward Dialing (DID) number assigned to you by your ITSP. Usually this is the same as your user ID, but it can be a different number. Check with your ITSP to find out what your DID number is.

Then decide which Internet phones will ring when an outside caller calls your DID number. The default is aa, which stands for auto-attendant, an automated system that picks up external calls and plays pre-recorded voice messages. If you want only the auto-attendant to receive a call, keep the default setting. When the auto-attendant receives a call, it will prompt the caller to dial the appropriate extension.

If you want specific Internet phones to ring when your DID number is called, then refer to “Chapter 6: Using the Web-based Utility” for instructions about the Contact List setting.

NOTE: If you decide to keep traditional phone service, which is also known as Plain Old Telephone Service (POTS), then you will use the Linksys Analog Telephone Adapter (model number: SPA3000). For details, refer to the Analog Telephone Adapter’s documentation.

Configuring the Auto-Attendant

By default, the daytime auto-attendant is enabled, so the first message it plays (“If you know your party’s extension, you may enter it now”) is suitable for business hours. If you want a caller to hear a different greeting during nighttime (non-business) hours, then refer to “Appendix B: Configuring the Nighttime Auto-Attendant.”

To use the Web-based Utility for additional configuration, refer to “Chapter 6: Using the Web-based Utility.” To use the Interactive Voice Response Menu, proceed to “Chapter 5: Using the Voice Interactive Response Menu.”

Chapter 5: Using the Interactive Voice Response Menu

Overview

You may need to manually configure the System by entering the settings provided by your Internet Telephony Service Provider (ITSP). This chapter explains how to use the Interactive Voice Response Menu to configure the System’s network settings and record auto-attendant messages. You will use the telephone’s keypad to enter your commands and select choices, and the System will use voice responses.

For more advanced configuration, refer to “Chapter 6: Using the Web-based Utility.”

NOTE: If your ITSP sent you the System, then it may be pre-configured for you, and you do not need to change any settings. Refer to the instructions supplied by your ITSP for more information.

Accessing the Interactive Voice Response Menu

  1. Use a telephone connected to the Phone 1 or Phone 2 port of the System. (You can only access the Interactive Voice Response Menu through an analog telephone, not any of the Internet phones.)
  2. Press **** (in other words, press the star key four times).
  3. Wait until you hear “Linksys configuration menu. Please enter the option followed by the # (pound) key or hang up to exit.”
  4. Refer to the following table that lists actions, commands, menu choices, and descriptions. After you select an option, press the # (pound) key. To exit the menu, hang up the telephone.

Using the Interactive Voice Response Menu

While entering a value, such as an IP address, you may exit without entering any changes. Press the * (star) key twice within half a second. Otherwise, the * will be treated as a decimal point or dot.

After entering a value, such as an IP address, press the # (pound) key to indicate you have finished your selection. To save the new setting, press 1. To review the new setting, press 2. To re-enter the new 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 these 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 System will time out. You will need to re-enter the menu by pressing ****.

The settings you have saved will take effect after you have hung up the telephone. The System may reboot at this time.

Interactive Voice Response Menu

Action Command (press these keys on the telephone) Choices Description
Enter Interactive Voice Response Menu **** Use this command to enter the Interactive Voice Response Menu. Do not press any other keys until you hear, “Linksys configuration menu. Please enter the option followed by the # (pound) key or hang up to exit.”
Check Internet Connection Type 100 Hear the Internet connection type of the System.
Check Internet IP Address 110 Hear the IP address assigned to the System’s Internet (external) interface.
Check Network Mask (or Subnet Mask) 120 Hear the network or subnet mask assigned to the System.
Check Gateway IP Address 130 Hear the IP address of the gateway (usually the network router).
Check MAC Address 140 Hear the MAC address of the System in hexadecimal string format.
Check Firmware Version 150 Hear the version number of the firmware currently running on the System.

ip (internet protocol): a protocol used to send data over a network.

ip address: the address used to identify a computer or device on a network.

subnet mask: an address code that determines the size of the network.

gateway: a device that forwards Internet traffic from your local area network.

mac address: the unique address that a manufacturer assigns to each networking device.

firmware: the programming code that runs a networking device.

Interactive Voice Response Menu

Action Command Choices Description
(press these keys on the
telephone)
Check Primary DNS 160 Hear the IP address of the primary
Server IP Address DNS (Domain Name Service) server.
Check Internet Web 170 Hear the port number of the Internet
Server Port Web server used for the Web-based
Utility.
Check Local IP 210 Hear the local IP address of the
Address System.
Set Internet 101 Press 0 to use DHCP. Select the type of Internet connection
Connection Type Press 1 to use a static IP you are using. Refer to the
address. documentation supplied by your
Press 2 to use PPPoE. Internet service provider.
Set Static IP Address 111 Enter the IP address First, set the Internet Connection Type
using numbers on the to static IP address; otherwise, you
telephone keypad. Use the * (star) key when will hear, “Invalid Option,” if you try to set the static IP address.
entering a decimal point.
Set Network (or 121 Enter the network or First, set the Internet Connection Type
Subnet) Mask subnet mask using to static IP address; otherwise, you
numbers on the telephone keypad. Use will hear, “Invalid Option,” if you try to set the network or subnet mask.
the * (star) key when
entering a decimal point.
Set Gateway IP 131 Enter the IP address First, set the Internet Connection Type
Address using numbers on the telephone keypad. Use to static IP address; otherwise, you will hear, “Invalid Option,” if you try to
the * (star) key when set the gateway IP address.
entering a decimal point.

dhcp (dynamic host configuration protocol): a protocol that lets one device on a local network, known as a DHCP server, assign temporary IP addresses to the other network devices, typically computers.

static ip address: a fixed address assigned to a computer or device that is connected to a network.

pppoe: a type of broadband connection that provides authentication (username and password) in addition to data transport.

Interactive Voice Response Menu

Action Command Choices Description
(press these keys on the
telephone)
Set Primary DNS Server IP Address 161 Enter the IP address using numbers on the First, set the Internet Connection Type to static IP address; otherwise, you
telephone keypad. Use the * (star) key when will hear, “Invalid Option,” if you try to set the IP address of the primary DNS
entering a decimal point. server.
Set the Mode 201 Press 0 to select the router/NAT mode. Press 1 to select the bridge/switch mode. Use the router/NAT mode when the Internet phones are on the Local Area Network (LAN) side.
Use the bridge/switch mode when the
Internet phones are on the Wide Area Network (WAN) side.
Configure 72255 Refer to the “Configuring the
Auto-Attendant Auto-Attendant Messages” section at
Messages the end of this chapter.
Enable/Disable WAN 7932 Press 1 to enable. Use this setting to enable or disable
Access to the Press 0 to disable. WAN access to the Web-based Utility.
Web-based Utility (This Utility lets you configure the
System.)
Manual Reboot 732668 After you hear, “Option successful,”
hang up the phone. The System will
automatically reboot.
Factory Reset 73738 Press 1 to confirm. Press * (star) to cancel. If necessary, enter the password. The System will request confirmation; enter 1 to confirm. You will hear,
“Option successful.” Hang up the
phone. The System will reboot, and all
settings will be reset to their factory
default settings.

NOTE: This feature may be protected by a password available only from your ITSP.

If you need to enter a password, refer to the following section, “Entering a Password.”

Interactive Voice Response Menu

Action Command Choices Description
(press these keys on the
telephone)
Change 79228 Press 0 to use the Use this setting to select the
Auto-Attendant auto-attendant based on auto-attendant you want to use. You
day and time. Press 1 to use the can have the auto-attendant change depending on the day and time, or you
Daytime Auto-Attendant. Press 2 to use the can use one auto-attendant for all days and hours. (Make sure the
Nighttime auto-attendant you select has been
Auto-Attendant. Press 3 to use the enabled through the Web-based Utility; otherwise, the auto-attendant
Weekend/Holiday feature will not work.)
Auto-Attendant.
For more information, refer to
“Chapter 6: Using the Web-based
Utility.”
User Factory Reset 877778 Press 1 to confirm. Press * (star) to cancel. The System will request confirmation; enter 1 to confirm. You will hear,
“Option successful.” Hang up the
phone. The System will reboot and all
user-configurable settings will be
reset to their factory default settings.

Entering a Password

You may be prompted to enter a password when you want to reset the System to its factory default settings. To enter the password, use the phone’s keypad, and follow the appropriate instructions.

  • To enter A, B, C, a, b, or c — press 2.
  • To enter D, E, F, d, e, or f — press 3.
  • To enter G, H, I, g, h, or i — press 4.
  • To enter J, K, L, j, k, or l — press 5.
  • To enter M, N, O, m, n, or o — press 6.
  • To enter P, Q, R, S, o, q, r, or s — press 7.
  • To enter T, U, V, t, u, or v — press 8.
  • To enter W, X, Y, Z, w, x, y, or z — press 9.
  • To enter all other characters, press 0.

NOTE: These bulleted instructions only apply when you are entering a password. At all other times, pressing a number only selects a number, not a letter or punctuation mark. For example, to enter the password phone@321 by keypad, press these keys: 746630321. Then press the # (pound) key to indicate that you have finished entering the password. To cancel your entry and return to the main menu, press * (star).

Configuring the Settings for Your Internet Phone Service

If you want to change the settings for your Internet phone service, refer to the instructions provided by your ITSP and “Chapter 6: Using the Web-based Utility.”

Configuring the Auto-Attendant Messages

The System provides a feature called the auto-attendant, which automatically answers incoming calls with greetings or directory messages. It can handle up to 10 incoming calls and uses the default user ID aa.

Auto-Attendant Messages

You can save up to 10 customized greetings. The first four have default messages, which can be changed through the Interactive Voice Response Menu.

Prompt ID Default Audio Message
1 “If you know your party’s extension, you may enter it now.”
2 “Your call has been forwarded.”
3 “Not a valid extension, please try again.”
4 “Goodbye.”

The recorded messages will be encoded with G711U and saved in flash memory. These messages will be erased whenever you reset the System to its factory default settings. The maximum length of any message is one minute. You can record up to 94.5 seconds of audio, excluding the default messages. When there is not enough memory left, the Interactive Voice Response Menu will automatically end the recording.

You can access the auto-attendant prompt settings through the Interactive Voice Response Menu.

  1. Using one of the analog telephones connected to the System, press **** (in other words, press the star key four times).
  2. Wait until you hear “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.
  4. You will hear, “Please enter the message number followed by the # (pound) key.” Enter the number of the message you wish to record, review, or delete.
  5. The Interactive Voice Response Menu will say, “Enter 1 to record. Enter 2 to review. Enter 3 to delete. Enter * to exit.” Follow the instructions for your selection.

1 to Record

a.
If you entered 1, you will hear, “You may record your message after the tone. When finished, press #.”
b.
After you record the message, you will hear, “To save, enter 1. To review, enter 2. To re-record, enter 3. To exit, enter *.”
c.
Follow the instructions for the entry you have selected. If you entered 1, the new message will be saved. You will be returned to the menu described in step 5. If you entered 2, you will hear the message played. You will be returned to the menu described in step b. If you entered 3, you will be returned to the menu in step a. If you entered *, you will be returned to the menu in step 5.

2 to Review If you entered 2, you will hear the message played. You will be returned to the menu described in step 5. 3 to Delete

a.
If you entered 3, you will hear, “Enter 1 to confirm; enter * to exit.
b.
If you entered 1, the message will be erased. You will be returned to the menu described in step 5. If you entered *, you will be returned to the previous menu described in step 5.

* to Exit

If you entered *, you will be returned to the previous menu in step 4.

Through the Web-based Utility, you can configure the auto-attendant to answer calls in a specific number of seconds. By default, the auto-attendant answer delay is set to 12 seconds for the daytime hours, while it is set to 0 seconds for nighttime hours and weekends.

For status information about the auto-attendant messages or to configure additional settings, such as the auto-attendant answer delay, refer to “Chapter 6: Using the Web-based Utility.”

NOTE: If there is not enough memory left to

record a new message, then you will hear,

“Option failed” and be returned to step 4.

NOTE: If the message you want to save is longer than 15 seconds, then you will hear, “One moment, please.” This indicates that it will take several seconds to save the message. After the message has been saved, you can continue to use the Interactive Voice Response Menu.

Chapter 6: Using the Web-based Utility

Overview

When you first install the System, Linksys strongly recommends that you use the Setup Wizard, which you can download from www.linksys.com. If you do not wish to run the Setup Wizard, you can use the Web-based Utility to configure the System.

The System may have been pre-configured by your Internet Telephony Service Provider (ITSP), so you may not have to make any changes. If you do wish to make changes, follow the instructions in this chapter.

The Web-based Utility offers two levels of access: user and admin (administrator). Your level of access depends on your service provider’s policies. Also, access to some settings may be protected or blocked, so that service settings cannot be accidentally changed. For more information, contact your ITSP.

This chapter will describe each web page of the Web-based Utility and each page’s key functions. The Internet connection settings are configured on the Router - WAN Setup screen, while some of the most popular features: auto-attendant, music-on-hold, and call hunt are configured on the Voice - SIP screen. The Utility can be accessed via your web browser through use of a computer on your network.

There are two main tabs: Router and Voice. Additional tabs will be available after you click one of the main tabs.

Router

  • Status. This screen displays routing information about the System.
  • WAN Setup. Use this screen to configure the Internet connection, MAC clone, remote management, QoS, VLAN, and optional settings.
  • LAN Setup. Use this screen to configure the local network, dynamic DHCP, and static DHCP lease settings.
  • Application. On this screen, configure port forwarding, DMZ, and reserved ports range settings. Voice
  • Info. This screen displays voice-related information about the System.
  • System. Use this screen to configure system settings. In most cases, you should not change these settings unless instructed to do by your ITSP.

NOTE: If you are not sure how to configure the settings, then keep the default settings.

  • SIP. Configure service, music-on-hold, group paging, call hunt, and auto-attendant settings on this screen. In most cases, do not change service settings unless instructed to do so by your ITSP.
  • Provisioning. Use this screen to configure service provisioning settings. In most cases, you should not change these settings unless instructed to do by your ITSP.
  • Regional. Use this screen to configure call settings. In most cases, you should not change these settings unless instructed to do by your ITSP.
  • FXS 1/2. Use the appropriate screen to configure settings for each FXS (Phone) port on the System.
  • Line 1/2/3/4. Use the appropriate screen to configure settings for each external Internet phone line.

How to Access the Web-based Utility

To access the Web-based Utility of the System, launch Internet Explorer or Netscape Navigator on the administration computer connected to the System’s Ethernet port. If the System uses its default address, then enter 192.168.0.1 in the Address field. If you have assigned a static IP address to the System, then enter <IP address of the System> in the Address field. Press the Enter key.

Enter your user name and password. The default user name for administrative access is admin, and the default user name for user access is user. (These user names cannot be changed.) Then enter the password supplied by your ITSP. (By default, there is no password, so if you were not given a password, then leave this field blank.)

To view the status information for the phones and their calls, click PBX Status. To switch to a different login, click User Login or Admin Login. Enter the appropriate login information. Two views of the Web-based Utility are available. Click basic to view basic settings, or click advanced to view advanced settings.

When you have finished making changes on a screen, click the Submit All Changes button to save the changes, or click the Undo All Changes button to undo your changes. When changes are saved, the System may reboot.

The PBX Status Screen

This screen shows status information for the phones and their calls.

Registration

This section shows the registration information for the phones.

Registration. To remove a phone’s registration, click its checkbox. Then click the Delete button.

NOTE: If your ITSP supplied the System, then it may be pre-configured for you, and you do not need to change any settings. Refer to the instructions supplied by your ITSP for more information.

Station. Shown here is the station name assigned to the phone. (This setting is configured through the phone.) User ID. Shown here is the extension number assigned to the phone. IP Address. Shown here is the local IP address of the phone. Reg Expires. This indicates the number of seconds left before the phone needs to re-register with the System.

Parking Lot

This section shows the calls that have been parked. Call park is a convenient feature that lets a call be put on hold and picked up from any extension number. Parking Lot. To remove a call from the Parking Lot, click its checkbox. Then click the Delete button. Caller ID. Shown here is the phone number of the caller. Parked By. Shown here is the extension number that parked the call.

Parked At. Shown here is the call park number that you should use to pick up this call.

Duration. Shown here is the length of time that the call has been parked. Line 1 Calls This section shows the current incoming and outgoing calls. Line 1 Calls. To remove a call, click its checkbox. Then click the Delete button. External. Shown here is the external phone number of the caller. Station. Shown here is the extension number of the call; it displays the word “callpark” when the call has been parked for pickup from any extension number. Figure 6-2: PBX Screen - Inbound Call Direction. Shown here is the direction of the call, Inbound or Outbound. State. Shown here is the status of the call, Connected or Proceeding. Duration. Shown here is the length of time the call has been active.

Chapter 6: Using the Web-based Utility The PBX Status Screen

The Router Tab

The Router - Status Screen This screen displays product and system information. Product Information Product Name. Shown here is the model number of the System. Serial Number. Shown here is the serial number of the System. Software Version. Shown here is the version number of the System software. Hardware Version. Shown here is the version number of the System hardware. MAC Address. Shown here is the MAC address of the System. Client Certificate. Shown here is the status of the client certificate. It authenticates the System for use in the

ITSP’s network.

Licenses. This indicates how many additional licenses you have acquired for the System. System Status Current Time. Displayed here is the current date and time of the System. Elapsed Time. Displayed here is the amount of time elapsed since the last reboot of the System. WAN Connection Type. Displayed here is the Internet connection type of the System. Current IP. Displayed here is the Internet IP address of the System. Host Name. Displayed here is the host name of the System. Domain. Displayed here is the domain name of the System. Current Netmask. Displayed here is the netmask or subnet mask of the System. Current Gateway. Displayed here is the IP address of the gateway. Primary DNS. Displayed here is the IP address of the primary DNS server.

mac address: the unique address that a manufacturer assigns to each networking device.

ip (internet protocol): a protocol used to send data over a network.

ip address: the address used to identify a computer or device on a network.

subnet mask: an address code that determines the size of the network.

gateway: a device that forwards Internet traffic from your local area network.

Secondary DNS. Displayed here is the IP address of the secondary DNS server. LAN IP Address. Displayed here is the local IP address of the System. Broadcast Pkts Sent. Displayed here is the number of broadcast packets sent. packet: a unit of data sent over a network. Broadcast Bytes Sent. Displayed here is the number of broadcast bytes sent. Broadcast Pkts Recv. Displayed here is the number of broadcast packets received and processed. Broadcast Bytes Recv. Displayed here is the number of broadcast bytes received and processed. Broadcast Pkts Dropped. Displayed here is the number of broadcast packets received but not processed. Broadcast Bytes Dropped. Displayed here is the number of broadcast bytes received but not processed.

The Router - WAN Setup Screen This screen lets you configure the Internet connection, MAC clone, remote management, QoS, VLAN, and optional

settings. Information about your Internet connection type should be provided by your Internet Service Provider (ISP). If you do not have this information, contact your service provider. Internet Connection Settings Connection Type. Select the connection type you use: DHCP, Static IP, or PPPOE. If you already have a router for your network, select Static IP and assign an address that is appropriate for your

network. (Refer to the router’s documentation for more information about IP addressing.) Static IP Settings If you selected Static IP, complete the Static IP Settings section.

Static IP. Enter the static or fixed IP address of the System (this should be provided by your ISP). NetMask. Enter the net or subnet mask of the System (this should be provided by your ISP). Gateway. Enter the IP address of the gateway (this should be provided by your ISP).

PPPOE Settings If you selected PPPOE, complete the PPPOE Settings section. PPPoE Login Name. Enter the name provided by your ISP. PPPOE Login Password. Enter the password provided by your ISP. PPPOE Service Name (optional). Enter the service name provided by your ISP. Optional Settings HostName. Enter the host name, if provided by your ISP. Domain. Enter the domain name, if provided by your ISP. Primary DNS. Enter the IP address of the primary DNS server. Secondary DNS (optional). Enter the IP address of the secondary DNS server.

dhcp (dynamic host configuration protocol): a protocol that lets one device on a local network, known as a DHCP server, assign temporary IP addresses to the other network devices, typically computers.

static ip address: a fixed address assigned to a computer or device that is connected to a network.

pppoe: a type of broadband connection that provides authentication (username and password) in addition to data transport

DNS Server Order. Select the order in which the DNS servers should be used: Manual; Manual, DHCP; or DHCP,

Manual. The default is Manual. DNS Query Mode. Select how the DNS servers should be queried: Parallel or Sequential. The default is Parallel.

Primary NTP Server. Enter the IP address of the primary NTP server, which the System uses to keep the date and time current. Secondary NTP Server (optional). Enter the IP address of the secondary NTP server.

MAC Clone Settings Enable MAC Clone Service. Select whether you want to clone a MAC address onto the System, yes or no. The default is no.

Cloned MAC Address. Enter the MAC address you want to clone. Remote Management Enable WAN Web Server. This feature lets you enable or disable access to the Web-based Utility from the WAN

side. Select yes or no from the drop-down menu. The default is no.

WAN Web Server Port. Enter the port number used to access the Utility from the WAN side. The default is 80. QOS Settings QOS QDisc. QoS prioritizes voice communications when different types of traffic are competing for bandwidth.

Select the method you want to use: NONE, CBQ, or TBF. The default is NONE.

Maximum Uplink Speed. Enter the maximum upload speed of your Internet connection. The default is 128Kbps. VLAN (Virtual Local Area Network) Settings Enable VLAN. VLAN (802.1Q) settings let you use the System in a virtual LAN environment. Select yes or no from

the drop-down menu. The default is no. VLAN ID. Enter the ID number used by the System. The default is 1. When you have finished making changes, click the Submit All Changes button to save the changes, or click the

Undo All Changes button to undo your changes.

The Router - LAN Setup Screen This screen lets you configure the local network, dynamic DHCP, and static DHCP lease settings. Networking Service. Select the service you want to use, NAT or Bridge. The default is NAT. LAN Network Settings LAN IP Address. Enter the local IP address of the System. The default is 192.168.0.1. LAN Subnet Mask. Select the local subnet mask: 255.255.255.0, 255.255.255.128, 255.255.255.192,

255.255.255.224, 255.255.255.240, 255.255.255.248, or 255.255.255.252. The default is 255.255.255.0. Enable DHCP Server. To use the System as a router assigning IP addresses, select yes. Otherwise, select no. The default is yes.

Figure 6-6: Router - LAN Setup Screen DHCP Lease Time. Enter the lease time used by the System to distribute IP addresses. The default is 24 Hours. DHCP Client Starting IP Address. When the System issues IP addresses, it starts with the first value of its DHCP client IP address range. Enter that value here. The default is 192.168.0.2. Number of Client IP Addresses. Enter the number of IP addresses that can be distributed. The default is 50. Static DHCP Lease Settings Enable. You can have the System assign the same IP address to a specific device. To disable this feature, select no. To use this feature, select yes. The default is no. Host MAC Address. Enter the MAC address of the device whose IP address you want to specify. Host IP Address. Enter the IP address you want to assign to the device, 192.168.0.x (x being a different number for each device you specify). When you have finished making changes, click the Submit All Changes button to save the changes, or click the Undo All Changes button to undo your changes.

The Router - Application Screen This screen lets you configure port forwarding, DMZ, and reserved ports range settings. Port Forwarding Settings Enable. Select yes or no for each port forwarding entry, which defines a port range to be forwarded to a server.

The default is no. Service Name. Enter the name of the service or application. Starting Port. Enter the starting port number of the forwarded port range. Ending Port. Enter the ending port number of the forwarded port range. Protocol. Select the protocol used, TCP, UDP, or Both. The default is TCP. Server IP Address. Enter the IP address of the server, 192.168.0.x (x being a different number for each server

you specify). DMZ Settings Enable DMZ. DMZ hosting forwards all ports at the same time to one computer. This allows one local user to be exposed to the Internet for use of special-purpose services such as videoconferencing. Select yes or no from the drop-down menu. The default is no. DMZ Host IP Address. Enter the IP address of the DMZ host, 192.168.0.x (x being the number for the computer

you want to specify). Use the Static DHCP Lease Settings section on the LAN Setup screen, so the DMZ Host keeps this IP address; otherwise, its IP address may change. System Reserved Ports Range

Starting Port. This port range defines the random TCP/UDP ports used by the application running on the System. They cannot be used by port forwarding or DMZ. Enter the starting port number of the reserved ports range. The default is 50000.

Num of Ports Reserved. Select the number of ports you want to reserve: 256, 512, or 1024. The default is 256. When you have finished making changes, click the Submit All Changes button to save the changes, or click the Undo All Changes button to undo your changes.

tcp: a network protocol for transmitting data that requires acknowledgement from the recipient of data sent.

udp: a network protocol for transmitting data that does not require acknowledgement from the recipient of the data that is sent.

The Voice Tab

The Voice - Info Screen This screen shows voice-related settings for the System.

Product Information Product Name. Shown here is the model number of the System. Serial Number. Shown here is the serial number of the System. Software Version. Shown here is the version number of the System software. Hardware Version. Shown here is the version number of the System hardware. MAC Address. Shown here is the MAC address of the System. Client Certificate. Shown here is the status of the client certificate, which indicates that the System has been

authorized by your ITSP.

Licenses. This indicates how many additional licenses you have acquired for the System. System Status Current Time. Displayed here is the current date and time of the System.

Figure 6-9: Voice - Info Screen - System Status Elapsed Time. Displayed here is the amount of time elapsed since the last reboot of the System. FXS 1/2 Status The FXS 1 and FXS 2 ports are the Phone ports of the System. (You can connect analog phones or fax machines to both ports.) They have the same status information available. Hook State. Displayed here is the status of the phone’s readiness. On indicates that the phone is ready for use, while Off indicates that the phone is in use. Message Waiting. This indicates whether you have new voicemail waiting. Call Back Active. This indicates whether a call back request is in progress. Last Called Number. Displayed here is the last number called.

Last Caller Number. Displayed here is the number of the last caller. Calls 1 and 2 have the same status information available. Call 1/2 State. Displayed here is the status of the call. Call 1/2 Tone. Displayed here is the type of tone used by the call. Call 1/2 Encoder. Displayed here is the codec used for encoding. Call 1/2 Decoder. Displayed here is the codec used for decoding. Call 1/2 FAX. Displayed here is the status of the fax pass-through mode. Call 1/2 Type. Displayed here is the direction of the call. Call 1/2 Remote Hold. This indicates whether the far end has placed the call on hold. Call 1/2 Callback. This indicates whether the call was triggered by a call back request. Call 1/2 Peer Name. Displayed here is the name of the internal phone. Call 1/2 Peer Phone. Displayed here is the phone number of the internal phone. Figure 6-10: Voice - Info Screen - FXS Status Call 1/2 Duration. Displayed here is the duration of the call. Call 1/2 Packets Sent. Displayed here is the number of packets sent. Call 1/2 Packets Recv. Displayed here is the number of packets received. Call 1/2 Bytes Sent. Displayed here is the number of bytes sent. Call 1/2 Bytes Recv. Displayed here is the number of bytes received. Call 1/2 Decode Latency. Displayed here is the number of milliseconds for decoder latency. Call 1/2 Jitter. Displayed here is the number of milliseconds for receiver jitter. Call 1/2 Round Trip Delay. Displayed here is the number of milliseconds for delay. Call 1/2 Packets Lost. Displayed here is the number of packets lost. Call 1/2 Packet Error. Displayed here is the number of invalid packets received.

Line 1/2/3/4 Status Lines 1, 2, 3, and 4 have the same status information available. Registration State. Shown here is the status of the line’s registration with the ITSP. Last Registration At. Shown here are the last date and time the line was registered. Next Registration In. Shown here is the number of seconds until the next registration.

Figure 6-11: Voice - Info Screen - Line Status Message Waiting. This indicates whether you have new voicemail waiting. Mapped SIP Port. Shown here is the port number of the mapped SIP port. Auto Attendant Prompt Status Prompt 1-4. The first four greetings or messages are defaults. If you change a default, then the screen will show the new prompt’s duration in milliseconds. Prompt 5-10. For each prompt, the screen shows its duration in milliseconds. Space Remaining. Shown here is the number of milliseconds available. Figure 6-12: Voice - Info Screen - Auto Attendant Prompt Status Current AA. Shown here is the auto-attendant in use. When you have finished making changes, click the Submit All Changes button to save the changes, or click the Undo All Changes button to undo your changes.

The Voice - System Screen This screen lets you configure system settings. IMPORTANT: In most cases, you should not change these settings unless instructed to do by your ITSP.

System Configuration

Restricted Access Domains. Enter the domain names permitted to access the System.

Enable Web Admin Access. This setting lets you enable or disable local access to the Web-based Utility. Select yes or no from the drop-down menu. The default is yes. Admin Passwd. Enter the password for the administrator. (By default, there is no password.) User Password. Enter the password for the user. (By default, there is no password.)

Miscellaneous Settings

Syslog Server. Enter the IP address of the syslog server, which logs system information and critical events of the System. Debug Server. Enter the IP address of the debug server, which logs debug information of the System. Debug Level. This determines the level of debug information that will be generated. Select 0, 1, 2, or 3 from the

drop-down menu. The higher the debug level, the more debug information will be generated. The default is 0,

which indicates that no debug information will be generated. When you have finished making changes, click the Submit All Changes button to save the changes, or click the Undo All Changes button to undo your changes.

The Voice - SIP Screen This screen lets you configure service, music-on-hold, group paging, call hunt, and auto-attendant settings. IMPORTANT: In most cases, you should not change the service settings unless instructed to do by your ITSP.

SIP Parameters Max Forward. This is the SIP Max Forward value, which can range from 1 to 255. The default is 70. Max Redirection. This is the number of times an invite can be redirected to avoid an infinite loop. The default

is 5. Max Auth. This is the maximum number of times (from 0 to 255) a request may be challenged. The default is 2. SIP User Agent Name. This is the User-Agent header used in outbound requests. The default is $VERSION. SIP Server Name. This is the Server header used in responses to inbound responses. The default is $VERSION. SIP Reg User Agent Name. This is the User-Agent name to be used in a REGISTER request. If this is not

specified, then the SIP User Agent Name will also be used for the REGISTER request.

SIP Accept Language. This is the Accept-Language header used by the System. There is no default (this indicates the System does not include this header). DTMF Relay MIME Type. This is the MIME Type used in a SIP INFO message to signal a DTMF event. The default

is application/dtmf-relay.

Hook Flash MIME Type. This is the MIME Type used in a SIP INFO message to signal a hook flash event. The default is application/hook-flash. Remove Last Reg. This feature lets you remove the last registration before registering a new one if the value is

different. Select yes or no from the drop-down menu. The default is no.

Use Compact Header. This feature lets you use compact SIP headers in outbound SIP messages. Select yes or no from the drop-down menu. The default is no. Escape Display Name. This feature lets you keep the Display Name private. Select yes if you want the System to

enclose the string (configured in the Display Name) in a pair of double quotes for outbound SIP messages. Any

occurrences of “ or \ in the string will be escaped with \” and \\ inside the pair of double quotes. Otherwise, select no. The default is no. SIP Timer Values (sec)

SIP T1. This is the RFC 3261 T1 value (RTT estimate), which can range from 0 to 64 seconds. The default is .5. SIP T2. This is the RFC 3261 T2 value (maximum retransmit interval for non-INVITE requests and INVITE responses), which can range from 0 to 64 seconds. The default is 4.

SIP T4. This is the RFC 3261 T4 value (maximum duration a message will remain in the network), which can range from 0 to 64 seconds. The default is 5. SIP Timer B. This is the INVITE time-out value, which can range from 0 to 64 seconds. The default is 32.

SIP Timer F. This is the non-INVITE time-out value, which can range from 0 to 64 seconds. The default is 32. SIP Timer H. This is the INVITE final response, time-out value, which can range from 0 to 64 seconds. The default is 32.

SIP Timer D. This is the ACK hang-around time, which can range from 0 to 64 seconds. The default is 32.

SIP Timer J. This is the non-INVITE response, hang-around time, which can range from 0 to 64 seconds. The default is 32. INVITE Expires. This is the INVITE request Expires header value. If you enter 0, then the Expires header is not

included in the request. The default is 240.

ReINVITE Expires. This is the ReINVITE request Expires header value. If you enter 0, then the Expires header is not included in the request. The default is 30. Reg Min Expires. This is the minimum registration expiration time allowed from the proxy in the Expires header

or as a Contact header parameter. If the proxy returns a value less than this setting, then the minimum value is

used. The default is 1. Reg Max Expires. This is the maximum registration expiration time allowed from the proxy in the Min-Expires header. If the value is larger than this setting, then the maximum value is used. The default is 7200.

Reg Retry Intvl. This is the interval to wait before the System retries registration after failing during the last registration. The default is 30.

Reg Retry Long Intvl. When registration fails with a SIP response code that does not match, the System will wait for the specified length of time before retrying. If this interval is 0, then the System will stop trying. This value should be much larger than the Reg Retry Intvl value. The default is 1200.

Response Status Code Handling SIT1-4 RSC. Enter the SIP response status code for the appropriate SIT Tone (SIT stands for Special Information

Tone). For example, if you set the SIT1 RSC to 404, then when the user makes a call and a failure code of 404 is Figure 6-16: Voice - SIP Screen - Response Status Code returned, the SIT1 tone is played. Handling Try Backup RSC. This is the SIP response code that retries a backup server for the current request. Retry Reg RSC. This is the interval to wait before the System retries registration after failing during the last

registration. RTP Parameters RTP Port Min. This is the minimum port number for RTP transmission and reception. The default is 16384. RTP Port Max. This is the maximum port number for RTP transmission and reception. The default is 16482. Figure 6-17: Voice - SIP Screen - RTP Parameters RTP Packet Size. This is the packet size in seconds, which can range from 0.01 to 0.16. Valid values must be a multiple of 0.01 seconds. The default is 0.030. Max RTP ICMP Err. This indicates that the RTP data stream has failed due to ICMP errors. The default is 0. RTCP Tx Interval. This is the interval for sending out RTCP sender reports on an active connection. It can range from 0 to 255 seconds. The default is 0. No UDP Checksum. Select yes if you want the System to calculate UDP header checksum for SIP messages. Otherwise, select no. The default is no. Stats in BYE. This sets whether the System will include the P-RTP-Stat header or response to a BYE message.

The header contains RTP statistics of the current call. Select yes or no from the drop-down menu. The default is no. SDP Payload Types

NSE Dynamic Payload. This is the NSE dynamic payload type. The default is 100. AVT Dynamic Payload. This is the AVT dynamic payload type. The default is 101. Figure 6-18: Voice - SIP Screen - SDP Payload Types

INFOREQ Dynamic Payload. This is the INFOREQ dynamic payload type. There is no default. G726r16 Dynamic Payload. This is the G726-16 dynamic payload type. The default is 98. G726r24 Dynamic Payload. This is the G726-24 dynamic payload type. The default is 97. G726r40 Dynamic Payload. This is the G726-40 dynamic payload type. The default is 96. G729b Dynamic Payload. This is the G729b dynamic payload type. The default is 99. NSE Codec Name. This is the NSE codec name used in SDP. The default is NSE. AVT Codec Name. This is the AVT codec name used in SDP. The default is telephone-event. G711u Codec Name. This is the G711u codec name used in SDP. The default is PCMU. G711a Codec Name. This is the G711a codec name used in SDP. The default is PCMA. G726r16 Codec Name. This is the G726-16 codec name used in SDP. The default is G726-16. G726r24 Codec Name. This is the G726-24 codec name used in SDP. The default is G726-24. G726r32 Codec Name. This is the G726-32 codec name used in SDP. The is G726-32. G726r40 Codec Name. This is the G726-40 codec name used in SDP. The default is G726-40. G729a Codec Name. This is the G729a codec name used in SDP. The default is G729a. G729b Codec Name. This is the G729b codec name used in SDP. The default is G729ab. G723 Codec Name. This is the G723 codec name used in SDP. The default is G723.

NAT Support Parameters Handle VIA received. If you select yes, the System will process the received parameter in the VIA header (this is inserted by the server in a response to any one of its requests). If you select no, the parameter will be ignored. Select yes or no from the drop-down menu. The default is no.

Insert VIA received. This lets you insert the received parameter into the VIA header of SIP responses if the received-from IP and VIA sent-by IP values differ. Select yes or no from the drop-down menu. The default is no.

Insert VIA rport. This feature lets you insert the rport parameter into the VIA header of SIP responses if the received-from port and VIA sent-by port numbers differ. Select yes or no from the drop-down menu. The default is no.

Substitute VIA Addr. This feature lets you use NAT-mapped IP:port values in the VIA header. Select yes or no from the drop-down menu. The default is no.

Send Resp To Src Port. This feature lets you send responses to the request source port instead of the VIA sent-by port. Select yes or no from the drop-down menu. The default is no.

STUN Enable. This feature lets you use STUN to discover NAT mapping. Select yes or no from the drop-down menu. The default is no.

STUN Test Enable. If the STUN Enable feature is enabled and a valid STUN server is available, then the System can perform a NAT type discovery operation when it powers on. It will contact the configured STUN server, and the result of the discovery will be reported in a Warning header in all subsequent REGISTER requests. If the System detects symmetric NAT or a symmetric firewall, NAT mapping will be disabled.

The STUN Test Enable feature lets you use the STUN test. Select yes or no from the drop-down menu. The default is no.

STUN Server. Enter the IP address of the STUN server to contact for NAT mapping discovery.

EXT IP. Enter the external IP address to substitute for the actual IP address of the System in all outgoing SIP messages. If 0.0.0.0 is specified, then no IP address substitution will be performed.

EXT RTP Port Min. This is the external port mapping number of the RTP Port Min. number. If this value is not zero, then the RTP port number in all outgoing SIP messages will be substituted for the corresponding port value in the external RTP port range.

NAT Keep Alive Intvl. This is the interval between NAT-mapping, keep alive messages. The default is 15.

PBX Parameters

Proxy Network Interface. This tells the System how the clients (usually phones) are connected. Select LAN or WAN. The default is WAN.

Proxy Listen Port. This is the port used by the System when it listens for client messages at the selected Figure 6-20: Voice - SIP Screen - PBX Parameters interface. The default is 6060.

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Multicast Address. This is the IP address (and port number) used by the System to send control messages to all clients at the same time. It must be a multicast address and must contain a port number. The default is 224.168.168.168:6061.

Group Page Address. This is the IP address (and port number) used by the System to tell clients to send and receive group page RTP packets. It must be a multicast address and must contain a port number. The default is 244.168.168.168:34567.

Max Expires. This sets the maximum allowed Registration Expires value (in seconds) for clients. The default is 3600.

Force Media Proxy. This feature forces external clients to use the System’s media proxy when exchanging RTP traffic with external peers. Select yes or no from the drop-down menu. The default is no.

Proxy Debug Option. SIP messages are received at or sent from the proxy listen port. This feature controls which SIP messages to log. Select none for no logging. Select 1-line to log the start-line only for all messages. Select 1-line excl. OPT to log the start-line only for all messages except OPTIONS requests/responses. Select 1-line excl. NTFY to log the start-line only for all messages except NOTIFY requests/responses. Select 1-line excl. REG to log the start-line only for all messages except REGISTER requests/responses. Select 1-line excl. OPT|NTFY|REG to log the start-line only for all messages except OPTIONS, NOTIFY, and REGISTER requests/responses. Select full to log all SIP messages in full text. Select full excl. OPT to log all SIP messages in full text except OPTIONS requests/responses. Select full excl. NTFY to log all SIP messages in full text except NOTIFY requests/responses. Select full excl. REG to log all SIP messages in full text except REGISTER requests/responses. Select full excl. OPT|NTFY|REG to log all SIP messages in full text except for OPTIONS, NOTIFY, and REGISTER requests/responses. The default is full.

Call Routing Rule. This is a special dial plan that determines which line can be used for an external, outbound call request from a phone based solely on the target public number. When you create this rule, follow this format:

(rule|rule|rule|...|rule)

The most specific rules should be placed first.

Each rule should be in this format: <:Lx>pattern

L indicates Line (phone line).

The variable x is 1, 2, 3, or 4 depending on which line you want to specify.

The word pattern indicates any digit pattern (see the Dial Plan setting for more information).

The default is (9xx.); this indicates that any of the four lines can be used for any target number starting with 9. For example, with this dial plan, the caller dials 9 before entering the external phone number.

Internal Music URL. Enter the Uniform Resource Locator (URL), also known as a web address, to download a music file for the music-on-hold and call park features. This is its format: [tftp://]server_IP_address[:port]/path. TFTP is the only protocol supported for music download. The default port is 69. Saving a new URL will reboot the System. After its reboot, the System will download the file and store the samples in flash memory.

The music samples are encoded in G711u format at 8000 samples/second. This file should not contain any extra header information, and its maximum length is 65.536 seconds (524,288 bytes). For more information, refer to “Appendix D: New Music for the Music-on-Hold Feature.”

Internal Music Script. This script tells the System how to play the downloaded music file. This is its format:

[section[,(section[,...]]]

Each section should be in this format: [n (start/end[/pause])] [pause2]

The variable n is the number of times you want a section to repeat before moving to the next section.

The start/end is the starting and 1+ending sample for the section. Note that the samples are numbered from 0 to total-length - 1. You may enter -1 or a very large number if the end of the file should be the ending sample. The default start value is 0, and the default end value is the end of the file.

The variable pause is the number of samples to pause after the ending sample has been played. The default is 0.

The variable pause2 is the additional number of samples to pause after the entire n repetitions of the section have been played. The default is 0.

A maximum of 16 sections can be specified. Samples should be encoded in G711u format at 8000 samples/second. After all sections are played, they are replayed, starting with the first section.

For example, the default Internal Music Script setting is 2(0/230954),2(230954/444720),(0/230954)40000. The first section is 2(0/230954); samples 0 through 230954 will be played twice. The second section is 2(230954/444720); samples 230954 through 444720 will be played twice. The third section is (0/230954); samples 0 through 230954 will be played once.

The last section is 40000. The ending pause will last for 40,000 samples. Each sample lasts 1/8000 of a second, so 40,000 samples equals 5 seconds. When this pause is over, the sections are replayed.

Internal MOH Refresh Intvl. The System can refresh an internal music session periodically. The default is 0, which disables the refresh function.

Call Park MOH Server. Enter the name or IP address of the music-on-hold server that should be used to handle a parked call. If you do not have a music-on-hold server for the call park feature, then keep the default, imusic, and the parked caller will hear the internal music file. Otherwise, if this setting is not specified, the parked caller will hear silence.

Call Park DLG Refresh Intvl. The System can refresh a call park session periodically. The default is 0, which disables the refresh function.

Default Group Line. This is the default group of lines, 1,2,3,4.

Group 1-4 User ID. A group designates specific phones that should be paged as a group, use the same phone lines, and receive the same type of calls. For example, sales calls should go to the sales group. You can designate up to four groups. For each group, enter a comma-separated list of User IDs, each representing a different client. For example, if the sales group is Group 1, then enter the sales extensions: 501,502,503 in the Group 1 User ID field. A client can belong to more than one group. If a client does not belong to any group, then the client belongs to the default group assigned to the Default Group Line. Each User ID pattern can use * and ? wildcards as well as %xx escaped characters (refer to “Appendix C: Dial Plan and Auto-Attendant Scripting for Advanced Users” for more details). The default is a blank field, which means that all clients belong to the default group.

Group 1-4 Line. For each group, enter a comma-separated list of phone lines the clients can use (this list determines the order in which the lines will be used). The System will make external calls for clients using the phone lines listed here. For example, for a group whose setting is 1,3, then System will use Line 1. If that is not successful, then it will use Line 3.

Hunt Groups. This defines one or more hunt groups that can be called directly by any client like a regular extension. The syntax is the same as the syntax for the Contact List. Note that a member of one group can also be the extension of another group (i.e., one level of recursion is allowed).

SIP DIDN Field. This determines which field is used to indicate the Direct Inward Dialing (DID) number for an incoming INVITE to a line interface. Select TO UserID to use the User-ID field of the TO header, or select TO Param to use a parameter in the TO header with the name specified in the SIP DIDN PARAM Name. The default is TO UserID.

SIP DIDN Param Name. This indicates the DID number in an incoming INVITE message. The default is didn. Auto Attendant Parameters AA Dial Plan 1. This is used to define the first dial rule in the auto-attendant. The default is (10x|xxx.). Refer to

“Appendix C: Dial Plan and Auto-Attendant Scripting for Advanced Users” for more details. AA Dial Plan 2. This is used to define the second dial rule in the auto-attendant. The default is (<:10>x|xxx.). Figure 6-21: Voice - SIP Screen - Auto Attendant

Parameters

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AA script 1-3. These are used to define the three auto-attendant scripts. Refer to “Appendix C: Dial Plan and Auto-Attendant Scripting for Advanced Users” for more details.

DayTime AA. To enable the daytime auto-attendant, select yes. Otherwise, select no. The default is yes. DayTime. Enter the daytime hours for the daytime auto-attendant in 24-hour format. Enter the start and end times in this format:

start=hh:mm:ss;end=hh:mm:ss (hh for hours, mm for minutes, and ss for seconds) For example, start=9:0:0;end=17:0:0 means the start time is 9 AM and the end time is 5 PM. The other hours

(5 PM to 9 AM) are considered nighttime hours.

If you do not enter start and end times, then the whole day (24 hours) is considered as daytime, so the nighttime auto-attendant will not be used, even if it is enabled. DayTime AA Script. Select the daytime auto-attendant script that you want to use, 1, 2, or 3. The default is 1. DayTime Answer Delay. Select the number of seconds you want the daytime auto-attendant to wait before

answering. The default is 12 seconds. NightTime AA. To enable the nighttime auto-attendant, select yes. Otherwise, select no. The default is no. NightTime AA Script. Select the nighttime auto-attendant script that you want to use, 1, 2, or 3. The default is 1. NightTime Answer Delay. Select the number of seconds you want the nighttime auto-attendant to wait before

answering. The default is 0 seconds. Weekend/Holiday AA. To enable this auto-attendant, select yes. Otherwise, select no. The default is no. Weekends/Holidays. When the weekend/holiday auto-attendant is enabled, you can use this setting to specify

the weekends and holidays. Up to four weekend days can be defined. Use this format: [wk=n1[,ni];][hd=mm/dd/yyyy|mm/dd/yyyy-mm/dd/yyyy[,mm/dd/yyyy|mm/dd/yyyy-mm/dd/yyyy];] (wk for weekend, which can be 1 for Monday to 7 for Sunday) (hd for holiday, which does not have to include the year) For example, wk=6,7;hd=1/1,2/21/2006,5/30/2006,12/19/2006-12/30/2006 means that Saturdays and Sundays

are the weekends. Holidays are January 1-2, 2006; May 30, 2006; and December 19-30, 2006.

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Weekend/Holiday AA Script. Select the auto-attendant script that you want to use for weekends and holidays, 1, 2, or 3. The default is 1.

Weekend/Holiday Answer Delay. Select the number of seconds you want the weekend/holiday auto-attendant to wait before answering. The default is 0 seconds.

PBX Phone Parameters

Next Auto User ID. This is the User ID assigned to the next new client that requests to register with the System.

Phone Ext Password. This is a REGISTRATION password that applies to Ext 1 of all clients. If there is no password, then all clients will be allowed to register without being challenged by the System. The default is blank Figure 6-22: Voice - SIP Screen - PBX Phone Parameters (no password).

Phone Upgrade Rule. This is the upgrade rule for all clients. The default is blank (no rule).

Phone Dial Plan. Enter the dial plan for all clients. The default is (9,[3469]11SO|9,<:1408>[2-9]XXXXXX|9,<:1>[2-9]xxxxxxxxxSO|9,1[2-9]xxxxxxxxxSO|9,011xx.|9,xx.|[1-8]xxx). This dial plan tells the phone to do the following:

  • play the outside dial tone if the first digit is 9
  • dial 9311, 9411, 9611, and 9911 immediately
  • dial 9 + [2-9] + 6 digits after a short timeout and insert the 1 + 408 area code
  • dial 9 + [2-9] + 9 digits immediately and insert the 1 (domestic long distance)
  • dial 91 + [2-9] + 9 digits immediately (domestic long distance)
  • dial 9011 + 1 or more digits after timeout or the # (pound) key (international)
  • dial 9 + 1 or more digits after timeout or the # (pound) key (catch-all)
  • dial [1-8] + 3 more digits immediately (internal calls)

Refer to “Appendix C: Dial Plan and Auto-Attendant Scripting for Advanced Users” for more details.

When you have finished making changes, click the Submit All Changes button to save the changes, or click the Undo All Changes button to undo your changes.

The Voice - Provisioning Screen

Use this screen to configure service provisioning settings.

IMPORTANT: In most cases, you should not change these settings unless instructed to do by your ITSP.

Configuration Profile

Provision Enable. The configuration profile must be requested by the System and cannot be pushed from a provisioning server, although a service provider can effectively push a profile by remotely triggering the request operation via SIP NOTIFY. To enable the provisioning feature, select yes. Otherwise, select no. The default is yes. Figure 6-23: Voice - Provisioning Screen - Configuration

Profile Resync On Reset. This feature lets you force the System to resync with the provisioning server when it powers on or reboots. Select yes or no from the drop-down menu. The default is yes.

Resync Random Delay. The System uses this feature to uniformly scatter resync requests from multiple devices over a period of time. Enter the period of time in seconds. The default is 2.

Resync Periodic. The System uses this feature to resync on a periodic basis. Enter the interval in seconds. The default is 3600.

Resync Error Retry Delay. If a resync attempt fails, the System will retry after a period of time. Enter the period of time in seconds. The default is 3600.

Forced Resync Delay. This feature tells the System how long to wait before undergoing a forced resync. Enter the period of time in seconds. The default is 14400.

Resync From SIP. This feature permits a service provider to trigger a profile resync via a SIP NOTIFY message. To enable this feature, select yes. Otherwise, select no. The default is yes.

Resync After Upgrade Attempt. If you want the System to resync after an upgrade attempt, select yes. Otherwise, select no. The default is yes.

Resync Trigger 1/2. Enter the first and second triggers you want to use.

Resync Fails On FNF. If you want the resync to fail when the FNF (File Not Found) error occurs, select yes. Otherwise, select no. The default is yes.

Profile Rule. This script identifies the provisioning server to contact when the System is performing a profile resync. Enter the appropriate script. The default is /spa$PSN.cfg.

Profile Rule B, C, and D. Enter profile rules B, C, and D.

Log Resync Request Msg. This script defines the message sent to the configured syslog server whenever the System attempts to resync with the provisioning server. Enter the appropriate script. The default is $PN $MAC -Requesting resync $SCHEME://$SERVIP:$PORT$PATH.

Log Resync Success Msg. This script defines the message sent to the configured syslog server whenever the System successfully completes a resync with the provisioning server. Enter the appropriate script. The default is $PN $MAC -- Successful resync $SCHEME://$SERVIP:$PORT$PATH.

Log Resync Failure Msg. This script defines the message sent to the configured syslog server whenever the System fails to complete a resync with the provisioning server. Enter the appropriate script. The default is $PN $MAC -- Resync failed: $ERR.

Report Rule. Enter the report rule.

Firmware Upgrade

Upgrade Enable. The firmware profile must be requested by the System and cannot be pushed from an upgrade server, although a service provider can effectively push a new firmware load by remotely triggering the request operation via the configuration file. To enable the upgrade feature, select yes. Otherwise, select no. The default is yes.

Upgrade Error Retry Delay. If an upgrade attempt fails, the System will retry after a period of time. Enter the period of time in seconds. The default is 3600.

Downgrade Rev Limit. Enter the downgrade firmware revision limit.

Upgrade Rule. Enter the upgrade rule.

Log Upgrade Request Msg. This script defines the message sent to the configured syslog server whenever the System attempts an upgrade from the upgrade server. Enter the appropriate script. The default is $PN $MAC -- Requesting upgrade $SCHEME://$SERVIP:$PORT$PATH.

Log Upgrade Success Msg. This script defines the message sent to the configured syslog server whenever the System successfully completes an upgrade from the upgrade server. Enter the appropriate script. The default is $PN $MAC -- successful upgrade $SCHEME://$SERVIP:$PORT$PATH -- $ERR.

Log Upgrade Failure Msg. This script defines the message sent to the configured syslog server whenever the System fails to complete an upgrade from the upgrade server. Enter the appropriate script. The default is $PN $MAC -- Upgrade failed $ERR.

firmware: the programming code

that runs a networking device.

upgrade: to replace existing software or

firmware with a newer version.

License Keys. There are additional license keys you can acquire to upgrade the System. You can upgrade from support of 4 phones to support of 16 phones, and/or you can upgrade from a two-line appearance per phone to a four-line appearance per phone. Enter the license keys in this field. For more information about licensing, contact your ITSP.

General Purpose Parameters

GPP A-P. These can be used by both the provisioning and upgrade logic to hold any string value. Then any of the values can be incorporated in other scripted parameters. Enter the appropriate string value in each field.

When you have finished making changes, click the Submit All Changes button to save the changes, or click the Undo All Changes button to undo your changes.

The Voice - Regional Screen

Use this screen to configure call settings.

IMPORTANT: In most cases, you should not change these settings unless instructed to do by your ITSP.

Call Progress Tones

Dial Tone. This is played to prompt the user to enter a phone number. The default is 350@-19,440@-19;10(*/0/ 1+2).

Second Dial Tone. This is an alternative to the Dial Tone when the user dials a three-way call. The default is 420@-19,520@-19;10(*/0/1+2).

Outside Dial Tone. This is an alternative to the Dial Tone. It prompts the user to enter an external phone number, as opposed to an internal extension. It is triggered by a , character (a comma) encountered in the dial plan. The default is 420@-19;10(*/0/1).

Prompt Tone. This is played to prompt the user to enter a call forwarding phone number. The default is 520@-19,620@-19;10(*/0/1+2).

Busy Tone. This is played when a 486 RSC is received for an outbound call. The default is 480@-19,620@-19;10(.5/.5/1+2).

Reorder Tone. This is played when an outbound call has failed or after the far end hangs up during an established call. The default is 480@-19,620@-19;10(.25/.25/1+2).

Off Hook Warning Tone. This is played when the caller has not properly placed the handset on the cradle. The default is 480@10,620@0;10(.125/.125/1+2).

Ring Back Tone. This is played during an outbound call when the far end is ringing. The default is 440@-19,480@-19;*(2/4/1+2).

Confirm Tone. This brief tone notifies the user that the last input value has been accepted. The default is 600@-16; 1(.25/.25/1).

SIT1 Tone. This is an alternative to the Reorder Tone played when an error occurs as a caller makes an outbound call. The RSC to trigger this tone is configurable on the SIP screen. The default is 985@-16,1428@-16,1777@-16;20(.380/0/1,.380/0/2,.380/0/3,0/4/0).

SIT2 Tone. This is an alternative to the Reorder Tone played when an error occurs as a caller makes an outbound call. The RSC to trigger this tone is configurable on the SIP screen. The default is 914@-16,1371@-16,1777@-16;20(.274/0/1,.274/0/2,.380/0/3,0/4/0).

SIT3 Tone. This is an alternative to the Reorder Tone played when an error occurs as a caller makes an outbound call. The RSC to trigger this tone is configurable on the SIP screen. The default is 914@-16,1371@-16,1777@-16;20(.380/0/1,.380/0/2,.380/0/3,0/4/0).

SIT4 Tone. This is an alternative to the Reorder Tone played when an error occurs as a caller makes an outbound call. The RSC to trigger this tone is configurable on the SIP screen. The default is 985@-16,1371@-16,1777@-16;20(.380/0/1,.274/0/2,.380/0/3,0/4/0).

MWI Dial Tone. This tone is played instead of the Dial Tone when there are unheard messages in the caller’s

mailbox. The default is 350@-19,440@-19;2(.1/.1/1+2);10(*/0/1+2). Cfwd Dial Tone. This tone is played when all calls are forwarded. The default is 350@-19,440@-19;2(.2/.2/ 1+2);10(*/0/1+2).

Holding Tone. This lets the local caller know that the far end has placed the call on hold. The default is 600@-

19*(.1/.1/1,.1/.1/1,.1/9.5/1). Conference Tone. This is played to all parties when a three-way conference call is in progress. The default is 350@-19;20(.1/.1/1,.1/9.7/1).

Secure Call Indication Tone. This is played when a call has been successfully switched to secure mode. It should be played only for a short while, less than 30 seconds, and at a reduced level, less than -19 dBm, so it will not interfere with the conversation. The default is 397@-19,507@-19;15(0/2/0,.2/.1/1,.1/2.1/2).

Feature Invocation Tone. This is played when a feature is implemented. The default is 350@-16;*(.1/.1/1). Distinctive Ring Patterns Ring1 Cadence. This is the cadence script for distinctive ring 1. The default is 60(2/4).

Ring2 Cadence. This is the cadence script for distinctive ring 2. The default is 60(.3/.2,1/.2,.3/4). Ring3 Cadence. This is the cadence script for distinctive ring 3. The default is 60(.8/.4,.8/4). Ring4 Cadence. This is the cadence script for distinctive ring 4. The default is 60(.4/.2,.3/.2,.8/4). Ring5 Cadence. This is the cadence script for distinctive ring 5. The default is 60(.2/.2,.2/.2,.2/.2,1/4).

Ring6 Cadence. This is the cadence script for distinctive ring 6. The default is 60(.2/.4,.2/.4,.2/4). Ring7 Cadence. This is the cadence script for distinctive ring 7. The default is 60(.4/.2,.4/.2,.4/4). Ring8 Cadence. This is the cadence script for distinctive ring 8. The default is 60(0.25/9.75).

Distinctive Call Waiting Tone Patterns CWT1 Cadence. This is the cadence script for distinctive CWT (Call Waiting Tone) 1. The default is 30(.3/9.7). CWT2 Cadence. This is the cadence script for distinctive CWT2. The default is 30(.1/.1, .1/9.7). CWT3 Cadence. This is the cadence script for distinctive CWT3. The default is 30(.1/.1, .3/.1, .1/9.3). CWT4 Cadence. This is the cadence script for distinctive CWT4. The default is 30(.1/.1,.1/.1,.1/9.5). CWT5 Cadence. This is the cadence script for distinctive CWT5. The default is 30(.3/.1,.1/.1,.3/9.1). CWT6 Cadence. This is the cadence script for distinctive CWT6. The default is 30(.1/.1,.3/.2,.3/9.1). CWT7 Cadence. This is the cadence script for distinctive CWT7. The default is 30(.3/.1,.3/.1,.1/9.1). CWT8 Cadence. This is the cadence script for distinctive CWT8. The default is 2.3(.3/2). Distinctive Ring/CWT Pattern Names Ring1 Name. In an INVITE’s Alert-Info header, this is the name that picks distinctive ring/CWT 1 for the inbound

call. The default is Bellcore-r1.

Ring2 Name. In an INVITE’s Alert-Info header, this is the name that picks distinctive ring/CWT 2 for the inbound call. The default is Bellcore-r2. Ring3 Name. In an INVITE’s Alert-Info header, this is the name that picks distinctive ring/CWT 3 for the inbound

call. The default is Bellcore-r3.

Ring4 Name. In an INVITE’s Alert-Info header, this is the name that picks distinctive ring/CWT 4 for the inbound call. The default is Bellcore-r4. Ring5 Name. In an INVITE’s Alert-Info header, this is the name that picks distinctive ring/CWT 5 for the inbound

call. The default is Bellcore-r5.

Ring6 Name. In an INVITE’s Alert-Info header, this is the name that picks distinctive ring/CWT 6 for the inbound

call. The default is Bellcore-r6. Ring7 Name. In an INVITE’s Alert-Info header, this is the name that picks distinctive ring/CWT 7 for the inbound call. The default is Bellcore-r7.

Ring8 Name. In an INVITE’s Alert-Info header, this is the name that picks distinctive ring/CWT 8 for the inbound call. The default is Bellcore-r8. Ring and Call Waiting Tone Spec Ring Waveform. Select the waveform of the ringing signal, Sinusoid or Trapezoid. The default is Sinusoid. Ring Frequency. Enter the frequency of the ringing signal, which can range from 10 to 100 Hz. The default is 25.

Ring Voltage. Enter the ringing voltage value, which can range from 60 to 90 volts. The default is 70. CWT Frequency. Enter the frequency script of the call waiting tone. All distinctive CWTs are based on this tone. The default is 440@-10.

Control Timer Values (sec)

Hook Flash Timer Min. This is the minimum on-hook time before off-hook to qualify as a hook-flash event. If the on-hook time is less than the minimum, then it is ignored. The range is 0.1 to 0.4 seconds. The default is .1. Hook Flash Timer Max. This is the maximum on-hook time before off-hook to qualify as a hook-flash event. If

the on-hook time is more than the maximum, then it is ignored. The range is 0.4 to 1.6 seconds. The default is .9.

Callee On Hook Delay. The phone must be on-hook for this length of time before the IP Telephony ends the current inbound call. (This does not apply to outbound calls.) The range is 0 to 255 seconds. The default is 0. Reorder Delay. This is the delay after the far end hangs up before the Reorder Tone is played. To play the tone

immediately, enter 0. To have the tone never play, enter inf. The range is 0 to 255 seconds. The default is 5.

Call Back Expires. This is the expiration time for a call back activation. The range is 0 to 65,535 seconds. The default is 1800. Call Back Retry Intvl. This is the interval for a call back retry. The range is 0 to 255 seconds. The default is 30. Call Back Delay. This is the delay between the System receiving the first SIP 18x response and the System

declaring that the far end is ringing. If a busy response is received during this time, then the System considers the call a failure and continues to retry. The default is .5.

Figure 6-30: Voice - Regional Screen - Ring and Call Waiting Tone Spec

VMWI Refresh Intvl. This is the interval between the VMWI refresh events to the CPE. The default is 0.

Interdigit Long Timer. This is the long timeout between entering digits when a caller is dialing. The range is 0 to 64 seconds. The default is 10. Interdigit Short Timer. This is the short timeout between entering digits when a caller is dialing. The range is 0

to 64 seconds. The default is 3. CPC Delay. CPC stands for Calling Party Control. The CPC Delay is the delay after the caller hangs up when the

System will begin removing the tip-and-ring voltage from the attached equipment of the called party. The range is 0 to 255 seconds, and the resolution is 1 second. The default is 2. CPC Duration. This is the length of time the tip-to-ring voltage is removed after the caller hangs up. After that,

the tip-to-ring voltage is restored, and the dial tone will apply if the attached equipment is still off-hook. The CPC is disabled if this value is set to 0. The range is 0 to 1.000 seconds, and the resolution is 0.001 seconds. The default is 0.

Vertical Service Activation Codes Call Return Code. This code calls the last caller. The default is *69. Call Redial Code. This code redials the last number called. The default is *07. Blind Transfer Code. This code begins a blind transfer of the current call to the extension specified after the

activation code. The default is *98. Call Back Act Code. This code starts a callback when the last outbound call is not busy. The default is *66. Call Back Deact Code. This code cancels a callback. The default is *86. Call Back Busy Act Code. This code starts a callback when the last outbound call is busy. The default is *05. Cfwd All Act Code. This code forwards all calls to the extension specified after the activation code. The default is Figure 6-32: Voice - Regional Screen - Vertical Service

*72. Activation Codes Cfwd All Deact Code. This code cancels call forwarding of all calls. The default is *73. Cfwd Busy Act Code. This code forwards busy calls to the extension specified after the activation code. The default is *90. Cfwd Busy Deact Code. This code cancels call forwarding of busy calls. The default is *91.

Cfwd No Ans Act Code. This code forwards no-answer calls to the extension specified after the activation code. The default is *92.

Cfwd No Ans Deact Code. This code cancels call forwarding of no-answer calls. The default is *93. Cfwd Last Act Code. This code forwards the last inbound or outbound calls to the extension specified after the activation code. The default is *63.

Cfwd Last Deact Code. This code cancels call forwarding of the last inbound or outbound calls. The default is *83. Block Last Act Code. This code blocks the last inbound call. The default is *60.

Block Last Deact Code. This code cancels blocking of the last inbound call. The default is *80. Accept Last Act Code. This code accepts the last outbound call. It lets the call ring through when the do not disturb or call forwarding of all calls features are enabled. The default is *64.

Accept Last Deact Code. This code cancels the code to accept the last outbound call. The default is *84. CW Act Code. This code enables call waiting on all calls. The default is *56. CW Deact Code. This code disables call waiting on all calls. The default is *57. CW Per Call Act Code. This code enables call waiting for the next call. The default is *71. CW Per Call Deact Code. This code disables call waiting for the next call. The default is *70. Block CID Act Code. This code blocks caller ID on all outbound calls. The default is *67. Block CID Deact Code. This code removes caller ID blocking on all outbound calls. The default is *68. Block CID Per Call Act Code. This code blocks caller ID on the next outbound call. The default is *81. Block CID Per Call Deact Code. This code removes caller ID blocking on the next inbound call. The default is

*82. Block ANC Act Code. This code blocks all anonymous calls. The default is *77. Block ANC Deact Code. This code removes blocking of all anonymous calls. The default is *87. DND Act Code. This code enables the do not disturb feature. The default is *78.

DND Deact Code. This code disables the do not disturb feature. The default is *79. CIC Act Code. This code enables caller ID generation. The default is *65. CIC Deact Code. This code disables caller ID generation. The default is *85. CWCID Act Code. This code enables call waiting, caller ID generation. The default is *25. CWCID Deact Code. This code disables call waiting, caller ID generation. The default is *45. Dist Ring Act Code. This code enables the distinctive ringing feature. The default is *26. Dist Ring Deact Code. This code disables the distinctive ringing feature. The default is *46. Speed Dial Act Code. This code assigns a speed dial number. The default is *74. Secure All Call Act Code. This code makes all outbound calls secure. The default is *16. Secure No Call Act Code. This code makes all outbound calls not secure. The default is *17. Secure One Call Act Code. This code makes the next outbound call secure. (It is redundant if all outbound calls

are secure by default.) The default is *18.

Secure One Call Deact Code. This code makes the next outbound call not secure. (It is redundant if all outbound calls are not secure by default.) The default is *19. Conference Act Code. If this code is specified, then the user must enter it before dialing the third party for a

conference call. Enter the code for a conference call.

Attn-Xfer Act Code. If the code is specified, then the user must enter it before dialing the third party for a call transfer. Enter the code for a call transfer. Modem Line Toggle Code. This code toggles the line to a modem. The default is *99. FAX Line Toggle Code. This code toggles the line to a fax machine. The default is #99. Referral Services Codes. These codes tell the System what to do when the user places the current call on hold

and is listening to the second dial tone. One or more * codes can be entered here. For example, the blind transfer code is *98. After the user dials *98, the System will wait for the user to enter a phone number. After the number has been entered, the System will complete the blind transfer for the call on hold.

Feature Dial Services Codes. These codes tell the System what to do when the user is listening to the first or second dial tone. One or more * codes can be entered here. For example, the code for call forwarding of all calls

Chapter 6: Using the Web-based Utility The Voice Tab

is *72. After the user dials *72, the System will wait for the user to enter a phone number. After the number has been entered, the System will forward all calls for that phone number. Vertical Service Announcement Codes Service Annc Base Number. Enter the base number for service announcements. Service Annc Extension Codes. Enter the extension codes for service announcements.

Figure 6-33: Voice -Regional Screen - Vertical Service Outbound Call Codec Selection Codes Announcement Codes

Prefer G711u Code. This is the dialing code that will make this codec the preferred codec for the associated call. The default is *017110.

Force G711u Code. This is the dialing code that will make this codec the only codec that can be used for the associated call. The default is *027110.

Prefer G711a Code. This is the dialing code that will make this codec the preferred codec for the associated call. The default is *017111. Figure 6-34: Voice - Regional Screen - Outbound Call Codec Selection Codes
Force G711a Code. This is the dialing code that will make this codec the only codec that can be used for the
associated call. The default is *027111.
Prefer G723 Code. This is the dialing code that will make this codec the preferred codec for the associated call.
The default is *01723.
Force G723 Code. This is the dialing code that will make this codec the only codec that can be used for the
associated call. The default is *02723.
Prefer G726r16 Code. This is the dialing code that will make this codec the preferred codec for the associated
call. The default is *0172616.
Force G726r16 Code. This is the dialing code that will make this codec the only codec that can be used for the
associated call. The default is *0272616.
Prefer G726r24 Code. This is the dialing code that will make this codec the preferred codec for the associated
call. The default is *0172624.
Force G726r24 Code. This is the dialing code that will make this codec the only codec that can be used for the
associated call. The default is *0272624.

Prefer G726r32 Code. This is the dialing code that will make this codec the preferred codec for the associated call. The default is *0172632.

Force G726r32 Code. This is the dialing code that will make this codec the only codec that can be used for the associated call. The default is *0272632.

Prefer G726r40 Code. This is the dialing code that will make this codec the preferred codec for the associated call. The default is *0172640.

Force G726r40 Code. This is the dialing code that will make this codec the only codec that can be used for the associated call. The default is *0272640.

Prefer G729a Code. This is the dialing code that will make this codec the preferred codec for the associated call. The default is *01729.

Force G729a Code. This is the dialing code that will make this codec the only codec that can be used for the associated call. The default is *02729.

Miscellaneous

Set Local Date (mm/dd). Set the local date (mm stands for months and dd stands for days). The year is optional and uses two or four digits.

Set Local Time (hh/mm). Set the local time (hh stands for hours and mm stands for minutes). Seconds are optional.

Time Zone. For caller ID generation, select the number of hours to add to GMT to generate the local time. The default is GMT-08:00.

FXS Port Impedance. This sets the electrical impedance of the FXS port. Select one of these choices: 600, 900, 600+2.16uF, 900+2.16uF, 270+750||150nF, 220+850||120nF, 220+820||115nF, or 370+620||310nF. The default is 600.

Daylight Saving Time Rule. Enter the rule for calculating daylight saving time; it should include the start, end, and save values. This rule is comprised of three fields. Each field is separated by ; (a semicolon) as shown below. Optional values inside [ ] (the brackets) are assumed to be 0 if they are not specified. Midnight is represented by

0:0:0 of the given date.

This is the format of the rule: Start = ; end=; save =

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

weekday> 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). The default Daylight Saving Time Rule is start=4/1/7;end=10/-1/7;save=1. FXS Port Input Gain. Enter the input gain in dB, up to three decimal places. The range is 6.0 to -infinity. The

default is -3.

FXS Port Output Gain. Enter the output gain in dB, up to three decimal places. The range is 6.0 to -infinity. The default is -3. DTMF Playback Level. Enter the local DTMF playback level in dBm, up to one decimal place. The default is -16. DTMF Playback Length. Enter the local DTMF playback duration in milliseconds. The default is .1. Detect ABCD. To enable local detection of DTMF ABCD, select yes. Otherwise, select no. The default is yes.

Playback ABCD. To enable local playback of OOB DTMF ABCD, select yes. Otherwise, select no. The default is yes.

Caller ID Method. You have a choice of methods to use for caller ID. Select Bellcore(N.Amer, China) for CID, CIDCW, and VMWI. FSK is sent after the first ring, and there is no polarity reversal or DTAS. Select DTMF(Finland,Sweden) for CID only. DTMF is sent after polarity reversal (with no DTAS) and before the first ring. Select DTMF(Denmark) for CID only. DTMF is sent after polarity reversal (with no DTAS) and before the first ring. Select ETSI DTMF for CID only. DTMF is sent after DTAS (with no polarity reversal) and before the first ring. Select ETSI DTMF With PR for CID only. DTMF is sent after polarity reversal and DTAS and before the first ring. Select ETSI DTMF After Ring for CID only. DTMF is sent after the first ring (with no polarity reversal or DTAS). Select ETSI FSK for CID, CIDCW, and VMWI. FSK is sent after DTAS (with no polarity reversal) and before the first ring. It will wait for ACK from CPE after DTAS for CIDCW. Select ETSI FSK With PR(UK) for CID, CIDCW, and VMWI. FSK is sent after polarity reversal and DTAS and before the first ring. It will wait for ACK from CPE after DTAS for CIDCW. Polarity reversal is applied only if the equipment is on-hook. The default is Bellcore(N.Amer, China).

Caller ID FSK Standard. The System supports bell 202 and v.23 standards for caller ID generation. Select the FSK standard you want to use, bell 202 or v.23. The default is bell 202.

Feature Invocation Method. Select the method you want to use, Default or Sweden default. The default is Default.

When you have finished making changes, click the Submit All Changes button to save the changes, or click the Undo All Changes button to undo your changes.

The Voice - FXS 1/2 Screen

Use the appropriate screen to configure settings for each FXS port, which is called the Phone port on the System.

IMPORTANT: In most cases, you should not change the service settings unless instructed to do by your ITSP.

Line Enable. To enable this line for service, select yes. Otherwise, select no. The default is yes.

Network Settings

SIP ToS/DiffServ Value. Enter the TOS/DiffServ field value in UDP IP packets carrying a SIP message. The default is 0x68.

SIP CoS Value. Enter the CoS value for SIP messages. The default is 3.

RTP ToS/DiffServ Value. Enter the ToS/DiffServ field value in UDP IP packets carrying RTP data. The default is 0xb8.

RTP CoS Value. Enter the CoS value for RTP data. The default is 6.

Network Jitter Level. This setting determines how jitter buffer size is adjusted by the System. Jitter buffer size is adjusted dynamically. The minimum jitter buffer size is 30 milliseconds or (10 milliseconds + current RTP frame size), whichever is larger, for all jitter level settings. However, the starting jitter buffer size value is larger for higher jitter levels. This setting controls the rate at which the jitter buffer size is adjusted to reach the minimum. Select the appropriate setting: low, medium, high, very high, or extremely high. The default is high.

Jitter Buffer Adjustment. This controls how the jitter buffer should be adjusted. Select the appropriate setting: up and down, up only, down only, or disable. The default is up and down.

SIP Settings

SIP Port. Enter the port number of the SIP message listening and transmission port. The default is 5080.

SIP Remote-Party-ID. To use the Remote-Party-ID header instead of the From header, select yes. Otherwise, select no. The default is yes.

Chapter 6: Using the Web-based Utility The Voice Tab

excl. NTFY to log the start-line only for all messages except NOTIFY requests/responses. Select 1-line excl. REG
to log the start-line only for all messages except REGISTER requests/responses. Select 1-line excl.
OPT|NTFY|REG to log the start-line only for all messages except OPTIONS, NOTIFY, and REGISTER
requests/responses. Select full to log all SIP messages in full text. Select full excl. OPT to log all SIP messages
in full text except OPTIONS requests/responses. Select full excl. NTFY to log all SIP messages in full text except
NOTIFY requests/responses. Select full excl. REG to log all SIP messages in full text except REGISTER
requests/responses. Select full excl. OPT|NTFY|REG to log all SIP messages in full text except for OPTIONS,
NOTIFY, and REGISTER requests/responses. The default is none.
RTP Log Intvl. Periodically, the System will log RTP statistics via syslog, depending on debug level. Enter the
period of time in seconds. The default is 0.
Restrict Source IP. If Lines 1 and 2 use the same SIP Port value and the Restrict Source IP feature is enabled,
then the proxy IP address for Lines 1 and 2 is treated as an acceptable IP address for both lines. To enable the
Restrict Source IP feature, select yes. Otherwise, select no. The default is no.
Referor Bye Delay. This controls when the System sends BYE to terminate stale call legs upon completion of call
transfers. Multiple delay settings (Referor, Refer Target, Referee, and Refer-To Target) are configured on this
screen. For the Referor Bye Delay, enter the appropriate period of time in seconds. The default is 4.
Refer Target Bye Delay. For the Refer Target Bye Delay, enter the appropriate period of time in seconds. The
default is 0.
Referee Bye Delay. For the Referee Bye Delay, enter the appropriate period of time in seconds. The default is 0.
Refer-To Target Contact. To contact the refer-to target, select yes. Otherwise, select no. The default is no.
Sticky 183. If this feature is enabled, then the IP Telephony will ignore further 180 SIP responses after receiving
the first 183 SIP response for an outbound INVITE. To enable this feature, select yes. Otherwise, select no. The
default is no.
Subscriber Information
Display Name. Enter the display name for caller ID.
User ID. Enter the extension number for this line. Figure 6-38: Voice - FXS 1 Screen - Subscriber
Information

Dial Plan

Dial Plan. Enter the dial plan script for this line. Refer to “Appendix C: Dial Plan and Auto-Attendant Scripting for Advanced Users” for more details.

Streaming Audio Server (SAS)

SAS Enable. To enable the use of the line as a streaming audio source, select yes. Otherwise, select no. If enabled, the line cannot be used for outgoing calls. Instead, it auto-answers incoming calls and streams audio RTP packets to the caller. The default is no.

SAS DLG Refresh Intvl. If this is not zero, it is the interval at which the streaming audio server sends out session refresh (SIP re-INVITE) messages to determine if the connection to the caller is still active. If the caller does not respond to the refresh message, then the System will end this call with a SIP BYE message. The range is 0 to 255 seconds (0 means that the session refresh is disabled).The default is 30.

SAS Inbound RTP Sink. This setting works around devices that do not play inbound RTP if the streaming audio server line declares itself as a send-only device and tells the client not to stream out audio. Enter a Fully Qualified Domain Name (FQDN) or IP address of an RTP sink; this will be used by the System’s streaming audio server line in the SDP of its 200 response to an inbound INVITE message from a client.

Call Feature Settings

Blind Attn-Xfer Enable. This settings lets the System perform an attended transfer operation by ending the current call leg and performing a blind transfer of the other call leg. If this feature is disabled, the System performs an attended transfer operation by referring the other call leg to the current call leg while maintaining both call legs. To use this feature, select yes. Otherwise, select no. The default is no.

MOH Server. Enter the user ID or URL of the auto-answering streaming audio server. When only a user ID is specified, the current or outbound proxy will be contacted. Music-on-hold is disabled if the MOH Server is not specified.

Xfer When Hangup Conf. This setting makes the System perform a transfer when a conference call has ended. Select yes or no from the drop-down menu. The default is yes.

Conference Bridge URL. This feature supports external conference bridging for n-way conference calls (n > 2), instead of mixing audio locally. To use this feature, set this parameter to that of the server's name, e.g., conf@myserver.com:12345 or conf (which uses the Proxy value as the domain).

Conference Bridge Ports. Select the maximum number of conference call participants. The range is 3 to 10. The default is 3.

Figure 6-39: Voice -FXS 1 Screen - Dial Plan

Figure 6-40: Voice -FXS 1 Screen - Streaming Audio Server

Enable IP Dialing. To use IP dialing, select yes. Otherwise, select no. The default is no.

Emergency Number. This is a comma-separated list of emergency number patterns. If the outbound call matches one of the patterns, then the System will disable hook flash event handling. Hook flash event handling will be restored to normal when the phone is on-hook again. If you leave this field blank, then the System will have no emergency number.

Mailbox ID. Enter the ID number of the mailbox for this line.

Audio Configuration

Preferred Codec. Select a preferred codec for all calls. (The actual codec used in a call still depends on the outcome of the codec negotiation protocol.) Select one of the following: G711u, G711a, G726-16, G726-24, G726-32, G726-40, G729a, or G723. The default is G711u.

Silence Supp Enable. To enable silence suppression so that silent audio frames are not transmitted, select yes. Otherwise, select no. The default is no.

Use Pref Codec Only. To only use the preferred codec for all calls, select yes. (The call will fail if the far end does not support this codec.) Otherwise, select no. The default is no.

Silence Threshold. Select the appropriate setting for the threshold: high, medium, or low. The default is medium.

G729a Enable. To enable the use of the G729a codec at 8kbps, select yes. Otherwise, select no. The default is yes.

Echo Canc Enable. To enable the use of the echo canceller, select yes. Otherwise, select no. The default is yes.

G723 Enable. To enable the use of the G723a codec at 6.3kbps, select yes. Otherwise, select no. The default is yes.

Echo Canc Adapt Enable. To enable the echo canceller to adapt, select yes. Otherwise, select no. The default is yes.

G726-16 Enable. To enable the use of the G726 codec at 16kbps, select yes. Otherwise, select no. The default is yes.

Echo Supp Enable. To enable the use of the echo suppressor, select yes. Otherwise, select no. The default is yes.

G726-24 Enable. To enable the use of the G726 codec at 24kbps, select yes. Otherwise, select no. The default is yes.

FAX CED Detect Enable. To enable detection of the fax Caller-Entered Digits (CED) tone, select yes. Otherwise, select no. The default is yes.

G726-32 Enable. To enable the use of the G726 codec at 32kbps, select yes. Otherwise, select no. The default is yes.

FAX CNG Detect Enable. To enable detection of the fax Calling Tone (CNG), select yes. Otherwise, select no. The default is yes.

G726-40 Enable. To enable the use of the G726 codec at 40kbps, select yes. Otherwise, select no. The default is yes.

FAX Passthru Codec. Select the codec for fax passthrough, G711u or G711a. The default is G711u.

DTMF Process INFO. To use the DTMF process info feature, select yes. Otherwise, select no. The default is yes.

FAX Codec Symmetric. To force the System to use a symmetric codec during fax passthrough, select yes. Otherwise, select no. The default is yes.

DTMF Process AVT. To use the DTMF process AVT feature, select yes. Otherwise, select no. The default is yes.

FAX Passthru Method. Select the fax passthrough method: None, NSE, or ReINVITE. The default is NSE.

DTMF Tx Method. Select the method to transmit DTMF signals to the far end: InBand, AVT, INFO, Auto, InBand+INFO, or AVT+INFO. InBand sends DTMF using the audio path. AVT sends DTMF as AVT events. INFO uses the SIP INFO method. Auto uses InBand or AVT based on the outcome of codec negotiation.The default is Auto.

FAX Process NSE. To use the fax process NSE feature, select yes. Otherwise, select no. The default is yes.

Hook Flash Tx Method. Select the method for signaling hook flash events: None, AVT, or INFO. None does not signal hook flash events. AVT uses RFC2833 AVT (event = 16). INFO uses SIP INFO with the single line signal=hf in the message body. The MIME type for this message body is taken from the Hook Flash MIME Type setting. The default is None.

FAX Disable ECAN. If enabled, this feature will automatically disable the echo canceller when a fax tone is detected. To use this feature, select yes. Otherwise, select no. The default is no.

Release Unused Codec. This feature allows the release of codecs not used after codec negotiation on the first call, so that other codecs can be used for the second line. To use this feature, select yes. Otherwise, select no. The default is yes.

FAX Enable T38. To enable the use of the ITU-T T.38 standard for faxing, select yes. Otherwise, select no. The default is yes.

FXS Port Polarity Configuration

Idle Polarity. Select the polarity before a call is connected, Forward or Reverse. The default is Forward.

Caller Conn Polarity. Select the polarity after an outbound call is connected, Forward or Reverse. The default is Figure 6-43: Voice - FXS 1 Screen - FXS Port Polarity Forward. Configuration Callee Conn Polarity. Select the polarity after an inbound call is connected, Forward or Reverse. The default is

Forward.

When you have finished making changes, click the Submit All Changes button to save the changes, or click the Undo All Changes button to undo your changes.

The Voice - Line 1/2/3/4 Screen

Use the appropriate screen to configure settings for each external Internet phone line.

IMPORTANT: In most cases, you should not change the service settings unless instructed to do by your ITSP. Figure 6-44: Voice - Line 1 Screen - Network Settings Line Enable. To enable this line for service, select yes. Otherwise, select no. The default is yes.

Network Settings

SIP ToS/DiffServ Value. Enter the TOS/DiffServ field value in UDP IP packets carrying SIP messages. The default is 0x68.

SIP CoS Value. Enter the CoS value for SIP messages. The default is 3.

SIP Settings

SIP Port. Enter the port number of the SIP message listening and transmission port. The default is 5060.

SIP 100REL Enable. To enable the support of 100REL SIP extension for reliable transmission of provisional responses (18x) and use of PRACK requests, select yes. Otherwise, select no. The default is no. Figure 6-45: Voice - Line 1 Screen - SIP Settings

Auth Resync-Reboot. If this feature is enabled, the System will authenticate the sender when it receives the NOTIFY resync reboot (RFC 2617) message. To use this feature, select yes. Otherwise, select no. The default is yes.

SIP Proxy-Require. The SIP proxy can support a specific extension or behavior when it sees this header from the user agent. If this field is configured and the proxy does not support it, then it will respond with the message, “unsupported.” Enter the appropriate header in the field provided.

SIP Remote-Party-ID. To use the Remote-Party-ID header instead of the From header, select yes. Otherwise, select no. The default is yes.

SIP Debug Option. SIP messages are received at or sent from the proxy listen port. This feature controls which SIP messages to log. Select none for no logging. Select 1-line to log the start-line only for all messages. Select 1-line excl. OPT to log the start-line only for all messages except OPTIONS requests/responses. Select 1-line excl. NTFY to log the start-line only for all messages except NOTIFY requests/responses. Select 1-line excl. REG to log the start-line only for all messages except REGISTER requests/responses. Select 1-line excl. OPT|NTFY|REG to log the start-line only for all messages except OPTIONS, NOTIFY, and REGISTER requests/responses. Select full to log all SIP messages in full text. Select full excl. OPT to log all SIP messages in full text except OPTIONS requests/responses. Select full excl. NTFY to log all SIP messages in full text except NOTIFY requests/responses. Select full excl. REG to log all SIP messages in full text except REGISTER requests/responses. Select full excl. OPT|NTFY|REG to log all SIP messages in full text except for OPTIONS, NOTIFY, and REGISTER requests/responses. The default is none.

Restrict Source IP. If Lines 1 and 2 use the same SIP Port value and the Restrict Source IP feature is enabled, then the proxy IP address for Lines 1 and 2 is treated as an acceptable IP address for both lines. To enable the Restrict Source IP feature, select yes. Otherwise, select no. The default is no.

Referor Bye Delay. This controls when the System sends BYE to terminate stale call legs upon completion of call transfers. Multiple delay settings (Referor, Refer Target, Referee, and Refer-To Target) are configured on this screen. For the Referor Bye Delay, enter the appropriate period of time in seconds. The default is 4.

Refer Target Bye Delay. For the Refer Target Bye Delay, enter the appropriate period of time in seconds. The default is 0.

Referee Bye Delay. For the Referee Bye Delay, enter the appropriate period of time in seconds. The default is 0.

Refer-To Target Contact. To contact the refer-to target, select yes. Otherwise, select no. The default is no.

Subscriber Information

Display Name. Enter the display name for caller ID.

User ID. Enter the extension number for this line.

Password. Enter the password for this line.

Use Auth ID. To use the authentication ID and password for SIP authentication, select yes. Otherwise, select no to use the user ID and password. The default is no.

Auth ID. Enter the authentication ID here.

Call Capacity. Select the maximum number of calls allowed on this line. (The System does not distinguish between incoming and outgoing calls when it determines its call capacity.)

Contact List. This is a list of clients that System should alert when there is an incoming call on this line. Each rule is also known as a hunt group. The default method to ring a group is to ring all the members simultaneously, unless a hunt rule is specified. The default contact list is aa (auto-attendant).

When you create this rule, follow this format: rule[|rule[|rule[...]]] The most specific rules should be placed first. Each rule should be in this format: [did:]ext[,ext[,ext[...]]][,name=gname][,hunt=hrule][,cfwd=target] The term did indicates an embedded Direct Inward Dialing (DID) number. If it is not specified, then the rule applies

to any DID number.

The term ext indicates the client extension number pattern. It accepts * and ? wildcards as well as %xx escaped characters. The term name is a name for the call group. If a hrule is specified, then the listed clients are contacted sequentially (also known as hunting); otherwise, they

will ring simultaneously. When you create this hrule, follow this format: hunt=<algo>;<interval>;<max> The term <algo> determines the order to ring the clients. It can be one of the following:

  • restart or re= so it always starts from the beginning of the list
  • next or ne= so it starts from the next on the list to the last client that rings
  • random or ra= so the order is random for each call

The term <interval> is the time, in seconds, to ring each client.

The term <max> is the total time, in seconds, to hunt before the call is rejected or forwarded to voicemail. If <max> is less than <interval>, it is interpreted as the number of cycles to go through the hunt group before call hunting stops. If <max> is 0, call hunting will continue indefinitely until the caller hangs up or someone answers the call.

If necessary, the call is forwarded to a user ID, called the target, in the hrule. If the target is a voicemail mailbox, then the target starts with vm. For example, the target vm3456 will have calls forwarded to voicemail with the mailbox ID 3456.

For example, the Contact List is 501,502,hunt=ne,4,1;cfwd=aa. This means that 501 will ring first for four seconds. If 501 does not pick up or is already on another call, then 502 will ring for four seconds. This cycle will repeat once before the call hunting stops. Then the call goes to the auto-attendant.

Cfwd No Ans Delay. Enter the delay, in seconds, before the call forwarding of no-answer calls feature is triggered. The default is 20.

Dial Plan Dial Plan. Enter the dial plan script for this line. Refer to “Appendix C: Dial Plan and Auto-Attendant Scripting for Advanced Users” for more details. The default is (<9:>xx.).

NAT Settings

NAT Mapping Enable. To use externally mapped IP addresses and SIP/RTP ports in SIP messages, select yes. Otherwise, select no. The default is no. NAT Keep Alive Enable. To send the configured NAT keep alive message periodically, select yes. Otherwise,

select no. The default is no. NAT Keep Alive Msg. Enter the keep alive message that should be sent periodically to maintain the current NAT

mapping. If the value is $NOTIFY, then a NOTIFY message is sent. If the value is $REGISTER, then a REGISTER message without contact is sent. The default is $NOTIFY. NAT Keep Alive Dest. Enter the destination that should receive NAT keep alive messages. If the value is $PROXY,

then the messages will be sent to the current or outbound proxy. The default is $PROXY. EXT SIP Port. Enter the port number of the external port to substitute for the actual SIP port of the System in all

outgoing SIP messages. Proxy and Registration Proxy. Enter the SIP proxy server for all outbound requests. Use Outbound Proxy. To use the outbound proxy, select yes. Otherwise, select no. The default is no. Outbound Proxy. Enter the SIP outbound proxy server, where all outbound requests are sent for the first hop. Use OB Proxy In Dialog. To force SIP requests to be sent to the outbound proxy within a dialog, select yes.

Otherwise, select no. The default is yes.

Register. To require periodic registration with the proxy server, select yes. Otherwise, select no. The default is yes. Make Call Without Reg. To allow outbound calls to be made without successful registration by the System,

select yes. Otherwise, select no. The default is no.

Figure 6-47: Voice - Line 1 Screen - Dial Plan

Register Expires. This is the expiration value, in seconds, of a REGISTER request. The System will periodically renew registration shortly before the current registration has expired. The default is 3600.

Ans Call Without Reg. To allow inbound calls to be answered without successful registration by the System, select yes. Otherwise, select no. The default is no.

Use DNS SRV. To use the DNS SRV lookup for proxy and outbound proxy, select yes. Otherwise, select no. The default is no.

DNS SRV Auto Prefix. To have the System automatically prepend the proxy or outbound proxy name with _sip._udp when it is performing a DNS SRV lookup on that name, select yes. Otherwise, select no. The default is no.

Proxy Fallback Intvl. This sets the delay, in seconds, after which the System will retry from the highest priority proxy (or outbound proxy) servers after it has failed over to a lower priority server. It will work only if the primary and backup proxy server list is provided to the System via the DNS SRV record lookup on the server name. The default is 3600.

Proxy Redundancy Method. The System will create an internal list of proxies returned in DNS SRV records. You have a choice of two modes. Select Normal mode if you want this list to contain proxies organized by weight and priority. Select Based on SRV Port mode if you want the System to use Normal mode first and then inspect the port number based on the first proxy’s port on the list. The default is Normal.

Mailbox Subscribe URL. Enter the URL that should receive the SUBSCRIBE messages, so the System will receive voicemail status notification for all mailboxes on this line.

Mailbox Deposit URL. Enter the URL that the System will contact when clients and external callers need to deposit voicemail in any of the mailboxes on this line.

Mailbox Manage URL. Enter the URL that the IP Telephony will contact when it needs to check voicemail for any of the mailboxes on this line.

Mailbox Status. Displayed here is the status for all mailboxes on this line. The status is automatically updated when the System receives voicemail status notification from the ITSP. The information is shown in this format:

[mailboxID:number of new messages/number of old messages[,mailboxID:number of new messages/number of old messages[,mailboxID:number of new messages/number of old messages[,...]]]]

When you have finished making changes, click the Submit All Changes button to save the changes, or click the Undo All Changes button to undo your changes.

Appendix A: Troubleshooting

This appendix provides solutions to problems that may occur during the installation and operation of the IP Telephony System. Read the description below to solve your problems. If you can't find an answer here, check the Linksys website at www.linksys.com.

Common Problems and Solutions

1. The System did not automatically assign an extension number to the Linksys SPA-family Internet phone, and the phone’s Ext LED is yellow instead of green.

Follow these steps:

A. Open the web browser on the administration computer.

B. Enter http://192.168.0.1/admin/router/status.

C. If the phone is on the WAN side, write down the Current IP of the System. (This is the Internet IP address.)

If the phone is on the LAN side, write down the LAN IP Address of the System. (This is the local IP

address.)

D. Access the phone’s Web-based Utility.

E. Make sure the configured proxy server on the phone matches the System’s IP address. (Refer to the phone’s documentation for details.)

2. The Internet phone can make internal calls to other Internet phones and analog phones; however, it cannot make external calls.

Check to see if the Internet phone’s line is registered. Follow these steps:

A. Open the web browser on the administration computer.

B. Enter http://192.168.0.1/admin/voice/advanced.

C. On the Voice - Info screen, check to see if the Line 1 Status indicates that the Registration State says, “Registered.”

D. If it is not registered, then verify that the User ID, Proxy, and Password supplied by your Internet Telephony Service Provider (ITSP) are valid (these settings are configured on the Line 1 screen).

3. I made a call from an outside line, and I did not hear a ring tone after I entered the extension number.

First, try again and make sure you entered the extension number correctly. If you still do not hear a ring tone, then follow these steps:

A. Open the web browser on the administration computer.

B. Enter http://192.168.0.1/admin/voice/status.

C. On the PBX Status screen, make sure the Internet phone for that extension number is registered.

4. I made a call from an outside line, and the auto-attendant says, “Not a valid extension, please try again.” However, I can make outgoing calls from the Internet phone with that extension number.

Follow these steps:

A. Open the web browser on the administration computer.

B. Enter http://192.168.0.1/admin/voice/advanced.

C. Click the SIP tab.

D. On the Voice - SIP screen, add the extension number to the Auto-Attendant dial plan.

5. When an outside line calls the System, it rings one time and then goes to the auto-attendant.

By default, if no one answers the call after four seconds, then the call will go to the auto-attendant. To change this setting, follow these steps:

A. Open the web browser on the administration computer.

B. Enter http://192.168.0.1/admin/voice/advanced.

C. Click the SIP tab.

D. On the Voice - SIP screen, change the appropriate Answer Delay setting (DayTime, NightTime, or Weekends/Holidays).

6. How can I change greetings for the auto-attendant?

Use the Interactive Voice Response Menu to record or change greetings; refer to “Chapter 5: Using the Interactive Voice Response Menu” for instructions.

7. I want to use another computer on the network (not the administration computer) to access the Web-based Utility. I entered http://192.168.0.1, but this address did not work.

Any computer connected to your router should use the Internet (WAN) IP address of the System. (The administration computer is directly connected to the System’s Ethernet port, so it can use http://192.168.0.1, which is the System’s local IP address.) Use the Interactive Voice Response Menu to find out the System’s Internet IP address. Follow these steps:

A. Use a telephone connected to the Phone 1 port of the System.

B. Press **** (in other words, press the star key four times).

C. Wait until you hear “Linksys configuration menu. Please enter the option followed by the # (pound) key or hang up to exit.”

D. Press 110#.

E. You will hear the IP address assigned to the System’s Internet (external) interface. Write it down.

F. Press 7932#.

G. Press 1 to enable WAN access to the Web-based Utility.

H. Open the web browser on a networked computer.

I. Enter http://(Internet IP address of the System).

8. I’m trying to access the System’s Web-based Utility, but I do not see the login screen. Instead, I see a screen saying, “404 Forbidden.”

If you are using Windows Explorer, perform the following steps until you see the Web-based Utility’s login screen (Netscape Navigator will require similar steps):

A. Click File. Make sure Work Offline is NOT checked.

B. Press CTRL + F5. This is a hard refresh, which will force Windows Explorer to load new webpages, not cached ones.

C. Click Tools. Click Internet Options. Click the Security tab. Click the Default level button. Make sure the security level is Medium or lower. Then click the OK button.

9. I need to set a static IP address on a PC.

The System, by default, assigns an IP address range of 192.168.0.100 to 192.168.0.150 using the DHCP server on the System. To set a static IP address, you can only use the ranges 192.168.0.2 to 192.168.0.99 and 192.168.0.151 to 192.168.0.254. Each PC or network device that uses TCP/IP must have a unique address to identify itself in a network. If the IP address is not unique to a network, Windows will generate an IP conflict error message. You can assign a static IP address to a PC by performing the following steps:

For Windows 98 and Millennium:

A. Click Start, Setting, and Control Panel. Double-click Network.

B. In The following network components are installed box, select the TCP/IP-> associated with your Ethernet adapter. If you only have one Ethernet adapter installed, you will only see one TCP/IP line with no association to an Ethernet adapter. Highlight it and click the Properties button.

C. In the TCP/IP properties window, select the IP address tab, and select Specify an IP address. Enter a unique IP address that is not used by any other computer on the network connected to the System. You can only use an IP address in the ranges 192.168.0.2 to 192.168.0.99 and 192.168.0.151 to 192.168.0.254. Make sure that each IP address is unique for each PC or network device.

D. Click the Gateway tab, and in the New Gateway prompt, enter 192.168.0.1, which is the default IP address of the System. Click the Add button to accept the entry.

E. Click the DNS tab, and make sure the DNS Enabled option is selected. Enter the Host and Domain names (e.g., John for Host and home for Domain). Enter the DNS entry provided by your ISP. If your ISP has not provided the DNS IP address, contact your ISP to get that information or go to its website for the information.

F. Click the OK button in the TCP/IP properties window, and click Close or the OK button for the Network window.

G. Restart the computer when asked.

For Windows 2000:

A. Click Start, Settings, and Control Panel. Double-click Network and Dial-Up Connections.

B. Right-click the Local Area Connection that is associated with the Ethernet adapter you are using, and select the Properties option.

C. In the Components checked are used by this connection box, highlight Internet Protocol (TCP/IP), and click the Properties button. Select Use the following IP address option.

D. Enter a unique IP address that is not used by any other computer on the network connected to the System. You can only use an IP address in the ranges 192.168.0.2 to 192.168.0.99 and 192.168.0.151 to 192.168.0.254.

E. Enter the Subnet Mask, 255.255.255.0.

F. Enter the Default Gateway, 192.168.0.1 (System’s default IP address).

G. Toward the bottom of the window, select Use the following DNS server addresses, and enter the Preferred DNS server and Alternative DNS server (provided by your ISP). Contact your ISP or go on its website to find the information.

H. Click the OK button in the Internet Protocol (TCP/IP) Properties window, and click the OK button in the Local Area Connection Properties window.

I. Restart the computer if asked.

For Windows XP: The following instructions assume you are running Windows XP with the default interface. If you are using the Classic interface (where the icons and menus look like previous Windows versions), please follow the instructions for Windows 2000.

A. Click Start and Control Panel.

B. Click the Network and Internet Connections icon and then the Network Connections icon.

C. Right-click the Local Area Connection that is associated with the Ethernet adapter you are using, and select the Properties option.

D. In the This connection uses the following items box, highlight Internet Protocol (TCP/IP). Click the Properties button.

E. Enter a unique IP address that is not used by any other computer on the network connected to the System. You can only use an IP address in the ranges 192.168.0.2 to 192.168.0.99 and 192.168.0.151 to 192.168.0.254.

F. Enter the Subnet Mask, 255.255.255.0.

G. Enter the Default Gateway, 192.168.0.1 (System’s default IP address).

H. Toward the bottom of the window, select Use the following DNS server addresses, and enter the Preferred DNS server and Alternative DNS server (provided by your ISP). Contact your ISP or go on its website to find the information.

I. Click the OK button in the Internet Protocol (TCP/IP) Properties window. Click the OK button in the Local Area Connection Properties window.

10. I want to test my Internet connection.

A. Check your TCP/IP settings.

For Windows 98 and Millennium: Refer to Windows Help for details. Make sure Obtain IP address automatically is selected in the settings.

For Windows 2000:

1. Click Start, Settings, and Control Panel. Double-click Network and Dial-Up Connections.

2. Right-click the Local Area Connection that is associated with the Ethernet adapter you are using, and select the Properties option.

3. In the Components checked are used by this connection box, highlight Internet Protocol (TCP/IP), and click the Properties button. Make sure that Obtain an IP address automatically and Obtain DNS server address automatically are selected.

  1. Click the OK button in the Internet Protocol (TCP/IP) Properties window, and click the OK button in the Local Area Connection Properties window.
  2. Restart the computer if asked.
  3. Click the OK button in the Internet Protocol (TCP/IP) Properties window, and click the OK button in the Local Area Connection Properties window.
  4. Restart the computer if asked.

For Windows XP: The following instructions assume you are running Windows XP with the default interface. If you are using the Classic interface (where the icons and menus look like previous Windows versions), please follow the instructions for Windows 2000.

  1. Click Start and Control Panel.
  2. Click the Network and Internet Connections icon and then the Network Connections icon.

3. Right-click the Local Area Connection that is associated with the Ethernet adapter you are using, and select the Properties option.

4. In the This connection uses the following items box, highlight Internet Protocol (TCP/IP), and click the Properties button. Make sure that Obtain an IP address automatically and Obtain DNS server address automatically are selected.

B. Open a command prompt.

  • For Windows 98 and Millennium, click Start and Run. In the Open field, type command. Press the Enter key or click the OK button.
  • For Windows 2000 and XP, click Start and Run. In the Open field, type cmd. Press the Enter key or click the OK button.

C. In the command prompt, type ping 192.168.0.1 and press the Enter key.

  • If you get a reply, the computer is communicating with the System.
  • If you do NOT get a reply, check the cable, and make sure Obtain an IP address automatically is selected in the TCP/IP settings for your Ethernet adapter.

D. In the command prompt, type ping followed by your Internet IP address and press the Enter key. The Internet IP Address can be found in the web interface of the IP Telephony System. For example, if your Internet IP address is 1.2.3.4, you would enter ping 1.2.3.4 and press the Enter key.

  • If you get a reply, the computer is connected to the System.
  • If you do NOT get a reply, try the ping command from a different computer to verify that your original computer is not the cause of the problem.

E. In the command prompt, type ping www.linksys.com and press the Enter key.

  • If you get a reply, the computer is connected to the Internet. If you cannot open a webpage, try the ping command from a different computer to verify that your original computer is not the cause of the problem.
  • If you do NOT get a reply, there may be a problem with the connection. Try the ping command from a different computer to verify that your original computer is not the cause of the problem.

11. I am not getting an IP address on the Internet with my Internet connection.

A. Refer to “Problem #10, I want to test my Internet connection” to verify that you have connectivity.

B. If you need to register the MAC address of your Ethernet adapter with your ISP, please see “Appendix E: Finding the MAC Address and IP Address for Your Ethernet Adapter.” If you need to clone the MAC address of your Ethernet adapter onto the System, see the Router - WAN Setup - MAC Clone Settings section of “Chapter 6: Using the Web-based Utility.”

C. Make sure you are using the right Internet settings. Contact your ISP to see if your Internet connection type is DHCP, Static IP Address, or PPPoE (commonly used by DSL consumers). Refer to the Router - WAN Setup - Internet Connection Settings section of “Chapter 6: Using the Web-based Utility.”

D. Make sure you use the right cable. Check to see if the LEDs for the Internet port are lit.

E. Make sure the cable connecting from your cable or DSL modem is connected to the Internet port of the System or your network router. Verify that the Router - Status page of the System’s Web-based Utility shows a valid IP address from your ISP.

F. Power off your network devices, including the System, cable/DSL modem, and router (if you have a separate router). Wait 30 seconds, and power on the cable/DSL modem first. Then power on the router (if used), System, and other network devices. Check the Router - Status tab of the System’s Web-based Utility to see if you get an IP address.

12. I am not able to access the System’s Web-based Utility Setup page.

A. Refer to “Problem #10, I want to test my Internet connection” to verify that your computer is properly connected to the System.

B. Refer to “Appendix E: Finding the MAC Address and IP Address for Your Ethernet Adapter” to verify that your computer has an IP Address, Subnet Mask, Gateway, and DNS.

C. Set a static IP address on your system; refer to “Problem #9: I need to set a static IP address on a PC.”

D. Refer to “Problem #17: I am a PPPoE user, and I need to remove the proxy settings or the dial-up pop-up window.”

13. I can’t get my Virtual Private Network (VPN) to work through the System.

VPNs that use IPSec with the ESP (Encapsulation Security Payload known as protocol 50) authentication will work fine. At least one IPSec session will work through the System; however, simultaneous IPSec sessions may be possible, depending on the specifics of your VPNs.

VPNs that use IPSec and AH (Authentication Header known as protocol 51) are incompatible with the System. AH has limitations due to occasional incompatibility with the NAT standard.

Change the IP address for the System to another subnet to avoid a conflict between the VPN IP address and your local IP address. For example, if your VPN server assigns an IP address 192.168.0.X (X is a number from 1 to 254) and your local LAN IP address is 192.168.0.X (X is the same number used in the VPN IP address), the System will have difficulties routing information to the right location. If you change the System’s IP address to 192.168.2.1, that should solve the problem. Change the System’s IP address through the Router - LAN Setup tab of the Web-based Utility. If you assigned a static IP address to any computer or network device on the network, you need to change its IP address accordingly to 192.168.2.Y (Y being any number from 1 to 254). Note that each IP address must be unique within the network.

Your VPN may require port 500/UDP packets to be passed to the computer that is connecting to the IPSec server. Refer to “Problem #15, I need to set up online game hosting or use other Internet applications” for details. Check the Linksys website at www.linksys.com for more information.

14. I need to set up a server behind my System.

To use a server like a web, ftp, or mail server, you need to know the respective port numbers they are using. For example, port 80 (HTTP) is used for web; port 21 (FTP) is used for FTP, and port 25 (SMTP outgoing) and port 110 (POP3 incoming) are used for the mail server. You can get more information by viewing the documentation provided with the server you installed. Follow these steps to set up port forwarding through the System’s Web-based Utility. We will be setting up web, ftp, and mail servers.

A. Access the System’s Web-based Utility by going to http://192.168.0.1 or the IP address of the System. Go to the Router => Application tab.

B. Select yes from the Enable drop-down menu.

C. Enter any name you want to use for the service.

D. Enter the port range of the service you are using. For example, if you have a web server, you would enter the range 80 (in the Starting Port field) to 80 (in the Ending Port field).

E. Select the protocol you will be using, TCP or UDP, or select Both.

F. Enter the IP address of the PC or network device that you want the port server to go to. For example, if the web server’s Ethernet adapter IP address is 192.168.0.100, you would enter 100 in the field provided. Check “Appendix E: Finding the MAC Address and IP Address for Your Ethernet Adapter” for details on getting an IP address.

G. Follow the instructions in steps B-F for the port services you want to use. Consider the examples below:

Enable Service Name Starting and Ending Ports Protocol IP Address
yes Web server 80 to 80 Both 192.168.0.100
yes FTP server 21 to 21 TCP 192.168.0.101
yes SMTP (outgoing) 25 to 25 Both 192.168.0.102
yes POP3 (incoming) 110 to 110 Both 192.168.0.102

When you have completed the configuration, click the Submit All Changes button.

15. I need to set up online game hosting or use other Internet applications.

If you want to play online games or use Internet applications, most will work without doing any port forwarding or DMZ hosting. There may be cases when you want to host an online game or Internet application. This would require you to set up the System to deliver incoming packets or data to a specific computer. This also applies to the Internet applications you are using. The best way to get the information on what port services to use is to go to the website of the online game or application you want to use. Follow these steps to set up online game hosting or use a certain Internet application:

A. Access the System’s Web-based Utility by going to http://192.168.0.1 or the IP address of the System. Go to the Router => Application tab.

B