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

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.

The Front Panel

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

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.”

Interactive Voice Response Menu

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

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

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

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.

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 #.”

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.”

* 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.

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.

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

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 - 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 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

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.

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:

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. Enter any name you want to use for the Application.

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

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

E. 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).

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

G. 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.

H. Check the Enabled option for the port services you want to use. Consider the examples below:

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

16. I can’t get the Internet game, server, or application to work.

If you are having difficulties getting any Internet game, server, or application to function properly, consider exposing one PC to the Internet using DeMilitarized Zone (DMZ) hosting. This option is available when an application requires too many ports or when you are not sure which port services to use. Make sure you disable all the forwarding entries if you want to successfully use DMZ hosting, since forwarding has priority over DMZ hosting. (In other words, data that enters the System will be checked first by the forwarding settings. If the port number that the data enters from does not have port forwarding, then the System will send the data to whichever PC or network device you set for DMZ hosting.) Follow these steps to set DMZ hosting:

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. Disable or remove the entries you have entered for forwarding. Keep this information in case you want to use it at a later time.

C. Select yes from the Enable DMZ drop-down menu.

D. Enter the Ethernet adapter’s IP address of the computer you want exposed to the Internet. This will bypass the NAT security for that computer. Please refer to “Appendix E: Finding the MAC Address and IP Address for Your Ethernet Adapter” for details on getting an IP address.

Once completed with the configuration, click the Submit All Changes button.

17. I am a PPPoE user, and I need to remove the proxy settings or the dial-up pop-up window.

If you have proxy settings, you need to disable these on your computer. Because the System is the gateway Internet access, the computer does not need proxy settings to gain access. Follow these directions to verify that you do not have any proxy settings and that the browser you use is set to connect directly to the LAN.

For Microsoft Internet Explorer 5.0 or higher:

A. Click Start, Settings, and Control Panel. Double-click Internet Options.

B. Click the Connections tab.

C. Click the LAN settings button and remove anything that is checked.

D. Click the OK button to go back to the previous screen.

E. Click the option Never dial a connection. This will remove any dial-up pop-ups for PPPoE users.

For Netscape 4.7 or higher:

A. Start Netscape Navigator, and click Edit, Preferences, Advanced, and Proxies.

B. Make sure you have Direct connection to the Internet selected on this screen.

C. Close all the windows to finish.

18. To start over, I need to set the System to its factory default settings.

Refer to “Chapter 5: Using the Interactive Voice Response Menu” for instructions.

19. I need to upgrade the firmware.

Firmware upgrades are handled by your ITSP. Contact your service provider for more information.

20. When I enter a URL or IP address, I get a time-out error or am prompted to retry.

Go through this checklist until your problem is solved:

• Check if other PCs work. If they do, ensure that your workstation’s IP settings are correct (IP Address, Subnet Mask, Default Gateway, and DNS). Restart the computer that is having a problem.
• If the PCs are configured correctly, but still not working, check the System. Make sure that it is connected and powered on. Check its settings. (If you cannot connect to it, check the LAN and power connections.)
• If you have a separate router, check the router. Make sure that it is connected and powered on. Check its settings. (If you cannot connect to it, check the LAN and power connections.)
• Check your Internet connection (DSL/cable modem) to see if it is working correctly. You can remove the System and router (if used) to verify a direct connection.
• Manually configure the TCP/IP with a DNS address provided by your ISP.
• Make sure that your browser is set to connect directly and that any dial-up is disabled. For Internet Explorer, click Tools, Internet Options, and then the Connection tab. Make sure that Internet Explorer is set to Never dial a connection. For Netscape Navigator, click Edit, Preferences, Advanced, and Proxy. Make sure that Netscape Navigator is set to Direct connection to the Internet.

Configuring the Auto-Attendant Settings

To configure the nighttime auto-attendant, follow these instructions:

1. Launch Internet Explorer or Netscape Navigator on the administration computer.
1. Enter /admin/voice/advanced in the web browser’s Address field. (Enter the IP address you assigned to the System when you installed it.)
2. Then press the Enter key.
2. The Voice - Info screen will appear. Click the SIP tab.
3. On the SIP screen, scroll down to the Auto Attendant Parameters.
4. For the AA script 2 field, follow these instructions:

a. Copy the default AA script 1 text to the Notepad (or other word processing program).

This is the default AA script 1 code:

> Figure B-2: Voice - SIP Screen - Auto Attendant

Parameters

Internal Dial Plan

When the auto-attendant is enabled, it parses and operates on user input (key presses or DTMF tones) following the rules specified in the auto-attendant dial plan of the System. These rules are specified by the AA Dial Plan parameters found on the Voice - SIP screen of the Web-based Utility.

Parameters of the Auto-Attendant Feature

The following parameters are the minimum that must be configured.

• Contact List. This is a list of clients that the System alerts when there is an incoming call on the line. The auto-attendant must be included on this list. By default, the auto-attendant is the only client on this list, so the auto-attendant will pick up every call. You can decide to have the auto-attendant pick up a call if a number (or group) of clients did not pick up the call first. This parameter is configured on the Voice - Line x (x is 1-4) screen of the Web-based Utility.
• AA Script. The System lets you program the auto-attendant instructions using XML scripting grammar. These parameters, AA Scripts 1-3, are configured on the Voice - SIP screen of the Web-based Utility. Only one script is active at a time. Scripting is described in more detail below.
• AA Dial Plan. The auto-attendant parses the user input according to one of the two parameters, AA Dial Plan 1 or 2. The AA Script includes a reference to one of these two dial plan parameters through the dial plan instruction. These parameters are configured on the Voice - SIP screen of the Web-based Utility.
• DayTime AA Script. This defines which of the three scripts (AA script 1, 2, or 3) should be used for daytime hours. This parameter is configured on the Voice - SIP screen of the Web-based Utility.

Prompts for the Auto-Attendant

The auto-attendant prompts are configured through the Interactive Voice Response Menu. To access these settings, follow these instructions:

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.

Refer to “Chapter 5: Using the Interactive Voice Response Menu” for more details.

The System can store up to 94.5 seconds of audio, excluding the default messages (Prompts 1-4). The maximum length of any message is one minute. The recorded messages will be encoded with G711U and saved in flash memory. Status information for these messages is found in the Auto-Attendant Prompt Status section of the Voice - Info screen of the Web-based Utility.

Each message is internally referred to as Prompt x, x being a number ranging from 1 to 10. You can customize the default prompts and add six additional prompts. When you reset the System to its factory default settings, customized messages will be erased, and prompts 1-4 will be restored to their default messages:

Table 3: Default Auto-Attendant Prompts

Customizing the Auto-Attendant

You have many parameters you can change to customize the auto-attendant. One of the most important parameters is the script, or set of instructions, that the auto-attendant executes when it is running. The next section will explain how to use XML scripting grammar.

The AA Scripts Parameters and XML Scripting

The System lets you use XML scripting grammar to define the auto-attendant instructions. You have a choice of three scripts, which are stored in the AA script 1-3 parameters on the Voice - SIP screen of the Web-based Utility. The instructions must be defined or encapsulated in a structure. You may have multiple structures within a script that the auto-attendant can transfer to, based on user input.

The XML scripting grammar supports two types of structures, node and menu. The main difference between the two types is that within the node type, user input cannot be processed—only actions may be specified. The format of the node type is as follows:

Appendix C: Dial Plan and Auto-Attendant Scripting for Advanced Users Configuring the Auto-Attendant

In the menu type, user input can be processed. It is processed according to the dial plan statement associated with the menu and defines what action the auto-attendant executes when the user input matches the dial plan. The format of the menu type is as follows:

The complete set of XML instructions are described in the following table:

Table 4: Auto-Attendant XML Instructions Set

Table 4: Auto-Attendant XML Instructions Set

Additional Parameters for Customization

Additional parameters are available on the Voice - SIP screen of the Web-based Utility and described in the following table:

Table 5: Additional Auto-Attendant Configuration Parameters