User manual LINKSYS SPA9000

IP Telephony System Voice User Guide Model No. SPA9000 IP Telephony System Copyright and Trademarks Specifications are subject to change without notice. Linksys is a registered trademark or trademark of Cisco Systems, Inc. and/or its affiliates in the U.S. and certain other countries. Copyright © 2006 Cisco Systems, Inc. All rights reserved. Other brands and product names are trademarks or registered trademarks of their respective holders. WARNING: This product contains chemicals, including lead, known to the State of California to cause cancer, and birth defects or other reproductive harm. Wash hands after handling. How to Use this Guide The guide to the IP Telephony System has been designed to make understanding networking with the IP Telephony System easier than ever. Look for the following items when reading this User Guide: This checkmark means there is a note of interest and is something you should pay special attention to while using the IP Telephony System. This exclamation point means there is a caution or warning and is something that could damage your property or the IP Telephony System. This question mark provides you with a reminder about something you might need to do while using the IP Telephony System. In addition to these symbols, there are definitions for technical terms that are presented like this: word: definition. Also, each figure (diagram, screenshot, or other image) is provided with a figure number and description, like this: Figure 0-1: Sample Figure Description Figure numbers and descriptions can also be found in the "List of Figures" section. SPA9000-UG-60613C DF IP Telephony System Table of Contents Chapter 1: Introduction Welcome What's in this Guide? 1 1 2 Chapter 2: Applications for the IP Telephony System How Does the IP Telephony System Fit into My Business or Home? What Does the IP Telephony System Do? A Typical Scenario Which Call Management Features Does the IP Telephony System Offer? 4 4 4 4 5 Chapter 3: Getting to Know the IP Telephony System The Back Panel The Front Panel 6 6 7 Chapter 4: Getting Started Overview Before You Begin Instructions for Installing the IP Telephony System Receiving and Handling External Phone Calls Configuring the Auto-Attendant 8 8 8 9 13 13 Chapter 5: Using the Interactive Voice Response Menu Overview Accessing the Interactive Voice Response Menu Using the Interactive Voice Response Menu Entering a Password Configuring the Settings for Your Internet Phone Service Configuring the Auto-Attendant Messages 14 14 14 14 19 19 20 Chapter 6: Using the Web-based Utility Overview How to Access the Web-based Utility The PBX Status Screen The Router Tab The Voice Tab 22 22 23 23 25 31 IP Telephony System Appendix A: Troubleshooting Common Problems and Solutions Frequently Asked Questions 70 70 80 Appendix B: Configuring the Nighttime Auto-Attendant Description of the Auto-Attendant Instructions for Setting Up the Nighttime Auto-Attendant 83 83 83 Appendix C: Dial Plan and Auto-Attendant Scripting for Advanced Users Overview Configuring Dial Plans Configuring Dial Plans for the Auto-Attendant Configuring the Auto-Attendant 87 87 87 89 89 Appendix D: New Music for the Music-on-Hold Feature Overview Before You Begin Instructions for Converting the Music File Instructions for Configuring the IP Telephony System 99 99 99 100 100 Appendix E: Finding the MAC Address and IP Address for Your Ethernet Adapter Windows 98 or Me Instructions Windows 2000 or XP Instructions For the Router's Web-based Utility 102 102 102 103 Appendix F: Windows Help Appendix G: Glossary Appendix H: Specifications Appendix I: Warranty Information Appendix J: Regulatory Information Appendix K: Contact Information Internet Telephony Service Provider (ITSP) Linksys 104 105 110 114 115 121 121 121 IP Telephony System List of Figures Figure 2-1: A Scenario for the IP Telephony System Figure 3-1: Back Panel Figure 3-2: Front Panel Figure 4-1: A Typical Scenario for the IP Telephony System Figure 4-2: Connect to the Phone 1 Port Figure 4-3: Connect to the Internet Port Figure 4-4: Connect to the Ethernet Port Figure 4-5: Connect to Power Figure 4-6: Voice - SIP Screen - PBX Parameters Figure 4-7: Router - WAN Setup Screen Figure 4-8: Voice - Line 1 Screen Figure 5-1: Auto-Attendant Options Figure 5-2: Auto-Attendant Message Options Figure 6-1: PBX Screen - Parking Lot Figure 6-2: PBX Screen - Inbound Call Figure 6-3: PBX Screen - Outbound Call Figure 6-4: Router - Status Screen Figure 6-5: Router - WAN Setup Screen Figure 6-6: Router - LAN Setup Screen Figure 6-7: Router - Application Screen Figure 6-8: Voice - Info Screen - Product Information Figure 6-9: Voice - Info Screen - System Status Figure 6-10: Voice - Info Screen - FXS Status Figure 6-11: Voice - Info Screen - Line Status Figure 6-12: Voice - Info Screen - Auto Attendant Prompt Status Figure 6-13: Voice - System Screen Figure 6-14: Voice - SIP Screen - SIP Parameters Figure 6-15: Voice - SIP Screen - SIP Timer Values 4 6 7 8 9 9 9 9 10 10 12 18 21 24 24 24 25 27 29 30 31 31 32 33 33 34 35 36 IP Telephony System Figure 6-16: Voice - SIP Screen - Response Status Code Handling Figure 6-17: Voice - SIP Screen - RTP Parameters Figure 6-18: Voice - SIP Screen - SDP Payload Types Figure 6-19: Voice - SIP Screen - NAT Support Parameters Figure 6-20: Voice - SIP Screen - PBX Parameters Figure 6-21: Voice - SIP Screen - Auto Attendant Parameters Figure 6-22: Voice - SIP Screen - PBX Phone Parameters Figure 6-23: Voice - Provisioning Screen - Configuration Profile Figure 6-24: Voice - Provisioning Screen - Firmware Upgrade Figure 6-25: Voice - Provisioning Screen - General Purpose Parameters Figure 6-26: Voice - Regional Screen - Call Progress Tones Figure 6-27: Voice - Regional Screen - Distinctive Ring Patterns Figure 6-28: Voice - Regional Screen - Distinctive Call Waiting Tone Patterns Figure 6-29: Voice - Regional Screen - Distinctive Ring/CWT Pattern Names Figure 6-30: Voice - Regional Screen - Ring and Call Waiting Tone Spec Figure 6-31: Voice - Regional Screen - Control Timer Values Figure 6-32: Voice - Regional Screen - Vertical Service Activation Codes Figure 6-33: Voice - Regional Screen - Vertical Service Announcement Codes Figure 6-34: Voice - Regional Screen - Outbound Call Codec Selection Codes Figure 6-35: Voice - Regional Screen - Miscellaneous Figure 6-36: Voice - FXS 1 Screen - Network Settings Figure 6-37: Voice - FXS 1 Screen - SIP Settings Figure 6-38: Voice - FXS 1 Screen - Subscriber Information Figure 6-39: Voice - FXS 1 Screen - Dial Plan Figure 6-40: Voice - FXS 1 Screen - Streaming Audio Server Figure 6-41: Voice - FXS 1 Screen - Call Feature Settings Figure 6-42: Voice - FXS 1 Screen - Audio Configuration Figure 6-43: Voice - FXS 1 Screen - FXS Port Polarity Configuration Figure 6-44: Voice - Line 1 Screen - Network Settings Figure 6-45: Voice - Line 1 Screen - SIP Settings 37 37 37 38 39 42 44 45 46 47 48 49 50 50 51 51 52 55 55 56 59 59 60 61 61 61 62 64 65 65 IP Telephony System Figure 6-46: Voice - Line 1 Screen - Subscriber Information Figure 6-47: Voice - Line 1 Screen - Dial Plan Figure 6-48: Voice - Line 1 Screen - NAT Settings Figure 6-49: Voice - Line 1 Screen - Proxy and Registration Figure B-1: Auto-Attendant Message Options Figure B-2: Voice - SIP Screen - Auto Attendant Parameters Figure E-1: IP Configuration Screen Figure E-2: MAC Address/Adapter Address Figure E-3: MAC Address/Physical Address Figure E-4: MAC Address Clone 66 68 68 68 84 85 102 102 103 103 IP Telephony System Chapter 1: Introduction Welcome Thank you for choosing the Linksys IP Telephony System. The System combines the rich feature set of legacy PBX (Private Branch eXchange) telephone systems with the convenience and cost advantages of Internet telephony. It supports common key system features such as an auto-attendant, music-on-hold, call forwarding, three-way call conferencing, and more. The System is so easy to configure that a fully working system can be set up in minutes. New Linksys SPA-family Internet telephones are automatically detected and registered when they are connected to the System. While the System will work with any SIP-compatible Internet telephone, it is the ideal host for Linksys business telephones, including model number: SPA941. The System supports the advanced features of these phones, such as shared line appearances, hunt groups, call transfer, call park, and group paging. Plus, with its two FXS ports, the System can support traditional analog devices such as telephones, fax machines, answering machines, media adapters. How does the System do all of this? By connecting your analog phones or fax machines to the System and connecting the System and Internet phones to your router, then the System can direct voice communications for your network. But what does all of this mean? Networks are useful tools for sharing Internet access and computer resources. Multiple computers can share Internet access, so you don't need more than one high-speed Internet connection. With Internet phone service, your Internet access can now be shared by your Internet phones as well. You will be able to make phone calls using your Internet phone service account, even while another colleague is web browsing. Plus, you can access one printer from different computers and access data located on another computer's hard drive (with the right permissions). PCs on a wired network create a LAN, or Local Area Network. They are connected with Ethernet cables, which is why the network is called "wired". The System takes your wired network and lets you integrate Internet phones and Internet phone service. When you first install the System, Linksys strongly recommends that you use the Setup Wizard, which you can download from www.linksys.com. If you do not wish to run the Setup Wizard, then use the instructions in the Quick Installation or this User Guide to help you. These instructions should be all you need to get the most out of the IP Telephony System. 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. 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. Chapter 1: Introduction Welcome 1 IP Telephony System 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. Chapter 1: Introduction What's in this Guide? 2 IP Telephony System · 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 1: Introduction What's in this Guide? 3 IP Telephony System 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 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. Internet 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 phone has its own extension number). Outgoing calls go to the correct external phone number (you can have more than one external phone number). 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 up to four ITSPs. To expand the basic configuration, contact your primary ITSP for more information. SPA941 SPA941 Switch Router Desktop Computer 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 those phones to make Internet phone or fax calls. (More details are available in "Chapter 4: Getting Started.") Analog Phone Fax Administration Computer Figure 2-1: A Scenario for the IP Telephony System Chapter 2: Applications for the IP Telephony System How Does the IP Telephony System Fit into My Business or Home? 4 IP Telephony System What Kind of Router Should I Use? For your network, get the highest-performance router possible. For best results, use a QoS (Quality of Service) router, so it can assign top priority to voice traffic. What Kind of Switch Should I Use? Again, performance is key. For best results, use a switch that offers QoS (Quality of Service) and full wire-speed switching. QoS enables the switch to give top priority to voice traffic, while full wire-speed switching lets it forward packets as fast as your network can deliver them. The next best choice is a switch featuring QoS (Quality of Service). What if I Keep My Traditional Phone Service? Traditional phone service, also known as Plain Old Telephone Service (POTS), runs on a network called the Public Switched Telephone Network (PSTN). If you decide to keep traditional phone service, then connect the Analog Telephone Adapter (model number: SPA3000) to the switch. (For more information, refer to the SPA3000 documentation.) Which Call Management Features Does the IP Telephony System Offer? Beyond basic call routing, the System offers several powerful and sophisticated features: · Auto-Attendant. An automated system guides each caller to the appropriate contact. · Music-on-Hold. You can combine the auto-attendant feature with the music- or information-on-hold feature, so the caller has a richer experience with your call system. · Call Hunt. You can designate which Internet phones receive outside calls. You can even have calls ring multiple phones, either simultaneously or one at a time. · Paging. When you want to page all of the Internet phones, you can use the System. · Dial Plans. When you have more than one dial plan, you can route outgoing calls to take advantage of the best rates available for the different types of calls. After setup of the System, you will have dynamic and feature-rich Internet voice communications for your business or home. NOTE: If your ITSP configured the System for you, then these features may already be set up. Check with your ITSP for more information. (To set up these features yourself, refer to "Chapter 6: Using the Web-based Utility.") Chapter 2: Applications for the IP Telephony System Which Call Management Features Does the IP Telephony System Offer? 5 IP Telephony System Chapter 3: Getting to Know the IP Telephony System The Back Panel The System's ports are located on its back panel. Figure 3-1: Back Panel PHONE 1/2 The PHONE 1/2 ports allow you to connect analog telephones (or fax machines) to the System using RJ-11 telephone cables (not included). The ETHERNET port connects to an administration computer, so you can access the System's Web-based Utility for configuration. This INTERNET port connects to either a router or broadband modem. The Power port is where you will connect the power adapter. ETHERNET INTERNET Power Chapter 3: Getting to Know the IP Telephony System The Back Panel 6 IP Telephony System The Front Panel The System's LEDs are located on its front panel. Figure 3-2: Front Panel Power Green. The power LED is solidly lit when the System is powered on and connected to the Internet. It flashes when there is no Internet connection. Green. The INTERNET LED is solidly lit when there is an Internet connection. It flashes when there is network activity. Green. The PHONE 1/2 LED is solidly lit when the phone is on-hook and registered. (The connection is registered if your Internet phone service account is active.) The LED is not lit when the phone is on-hook and not registered. It flashes when the phone is off-hook. INTERNET PHONE 1/2 Chapter 3: Getting to Know the IP Telephony System The Front Panel 7 IP Telephony System 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 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." Internet Before You Begin 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) · 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 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 Phone Fax Administration Computer SPA941 Switch Router Desktop Computer SPA941 Cable/DSL Modem SPA941 Figure 4-1: A Typical Scenario for the IP Telephony System Chapter 4: Getting Started Overview 8 IP Telephony System Instructions for Installing the IP Telephony System Internal Calls To install the System for internal calls, you will do the following: · connect and configure the System · connect the Internet phones 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. Connect and Configure the System 1. (optional) Plug an analog telephone into the Phone 1 port of the System. 2. (optional) If you have a second analog telephone or fax machine, plug it into the Phone 2 port of the System. 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. 3. 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. 4. Connect a different Ethernet network cable to the Ethernet port of the System. Then connect the other end to the computer you will use to manage the System (this will be called the administration computer). 5. Connect the included power adapter to the Power port of the System, and then plug the power adapter into an electrical outlet. 6. 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 4-3: Connect to the Internet Port Figure 4-2: Connect to the Phone 1 Port Figure 4-4: Connect to the Ethernet Port Figure 4-5: Connect to Power Chapter 4: Getting Started Instructions for Installing the IP Telephony System 9 IP Telephony System 8. The Voice - Info screen will appear. Click the SIP tab. 9. In the PBX Parameters section, select WAN from the Proxy Network Interface drop-down menu. 10. Click the Submit All Changes button. 11. The Voice - Info screen will appear. Click the Router tab. 12. Click the WAN Setup tab. 13. From the Connection Type drop-down menu, select Static IP. 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. 15. In the Optional Settings section, complete the Primary DNS field. Primary DNS. Enter the DNS IP address of your network router. 16. In the Remote Management section, select yes from the Enable WAN Web Server drop-down menu. 17. Click the Submit All Changes button. Figure 4-7: Router - WAN Setup Screen Figure 4-6: Voice - SIP Screen - PBX Parameters Chapter 4: Getting Started Instructions for Installing the IP Telephony System 10 IP Telephony System 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. 3. Connect an Ethernet network cable to an Internet phone. Then connect the other end to one of the Ethernet ports on the switch. (If the Internet phone has been used before, reset it to its factory default settings first. Refer to its documentation for more information.) 4. Connect the Internet phone's power adapter to its power port, and then plug the power adapter into an electrical outlet. 5. 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 model number SPA941). If you connect a different SIP-compatible phone, then registration 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. NOTE: The default SIP port of the System is 6060. Chapter 4: Getting Started Instructions for Installing the IP Telephony System 11 IP Telephony System 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 /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. 5. Click the Submit All Changes button to save your new settings. 6. The System will reboot itself. Then the Internet phones will reboot themselves. 7. 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. Figure 4-8: Voice - Line 1 Screen 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. 12 Chapter 4: Getting Started Instructions for Installing the IP Telephony System IP Telephony System Receiving and Handling External Phone Calls To receive external phone calls, you need to know the Direct Inward Dialing (DID) number assigned to you by your ITSP. Usually this is the same as your user ID, but it can be a different number. Check with your ITSP to find out what your DID number is. Then decide which Internet phones will ring when an outside caller calls your DID number. The default is aa, which stands for auto-attendant, an automated system that picks up external calls and plays pre-recorded voice messages. If you want only the auto-attendant to receive a call, keep the default setting. When the auto-attendant receives a call, it will prompt the caller to dial the appropriate extension. If you want specific Internet phones to ring when your DID number is called, then refer to "Chapter 6: Using the Web-based Utility" for instructions about the Contact List setting. NOTE: If you decide to keep traditional phone service, which is also known as Plain Old Telephone Service (POTS), then you will use the Linksys Analog Telephone Adapter (model number: SPA3000). For details, refer to the Analog Telephone Adapter's documentation. Configuring the Auto-Attendant By default, the daytime auto-attendant is enabled, so the first message it plays ("If you know your party's extension, you may enter it now") is suitable for business hours. If you want a caller to hear a different greeting during nighttime (non-business) hours, then refer to "Appendix B: Configuring the Nighttime Auto-Attendant." To use the Web-based Utility for additional configuration, refer to "Chapter 6: Using the Web-based Utility." To use the Interactive Voice Response Menu, proceed to "Chapter 5: Using the Voice Interactive Response Menu." Chapter 4: Getting Started Receiving and Handling External Phone Calls 13 IP Telephony System Chapter 5: Using the Interactive Voice Response Menu Overview You may need to manually configure the System by entering the settings provided by your Internet Telephony Service Provider (ITSP). This chapter explains how to use the Interactive Voice Response Menu to configure the System's network settings and record auto-attendant messages. You will use the telephone's keypad to enter your commands and select choices, and the System will use voice responses. For more advanced configuration, refer to "Chapter 6: Using the Web-based Utility." NOTE: If your ITSP sent you the System, then it may be pre-configured for you, and you do not need to change any settings. Refer to the instructions supplied by your ITSP for more information. Accessing the Interactive Voice Response Menu 1. Use a telephone connected to the Phone 1 or Phone 2 port of the System. (You can only access the Interactive Voice Response Menu through an analog telephone, not any of the Internet phones.) 2. Press **** (in other words, press the star key four times). 3. Wait until you hear "Linksys configuration menu. Please enter the option followed by the # (pound) key or hang up to exit." 4. Refer to the following table that lists actions, commands, menu choices, and descriptions. After you select an option, press the # (pound) key. To exit the menu, hang up the telephone. Using the Interactive Voice Response Menu While entering a value, such as an IP address, you may exit without entering any changes. Press the * (star) key twice within half a second. Otherwise, the * will be treated as a decimal point or dot. After entering a value, such as an IP address, press the # (pound) key to indicate you have finished your selection. To save the new setting, press 1. To review the new setting, press 2. To re-enter the new setting, press 3. To cancel your entry and return to the main menu, press * (star). Chapter 5: Using the Interactive Voice Response Menu Overview 14 IP Telephony System For example, to enter the IP address 191.168.1.105 by keypad, press these keys: 191*168*1*105. Press the # (pound) key to indicate that you have finished entering the IP address. Then press 1 to save the IP address or press the * (star) key to cancel your entry and return to the main menu. If the menu is inactive for more than one minute, the System will time out. You will need to re-enter the menu by pressing ****. The settings you have saved will take effect after you have hung up the telephone. The System may reboot at this time. Interactive Voice Response Menu Action Command (press these keys on the telephone) **** Choices Description Enter Interactive Voice Response Menu Use this command to enter the Interactive Voice Response Menu. Do not press any other keys until you hear, "Linksys configuration menu. Please enter the option followed by the # (pound) key or hang up to exit." Hear the Internet connection type of the System. Hear the IP address assigned to the System's Internet (external) interface. Hear the network or subnet mask assigned to the System. Hear the IP address of the gateway (usually the network router). Hear the MAC address of the System in hexadecimal string format. Hear the version number of the firmware currently running on the System. Check Internet Connection Type Check Internet IP Address Check Network Mask (or Subnet Mask) Check Gateway IP Address Check MAC Address Check Firmware Version 100 110 120 130 140 150 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. 15 Chapter 5: Using the Interactive Voice Response Menu Using the Interactive Voice Response Menu IP Telephony System Interactive Voice Response Menu Action Command (press these keys on the telephone) 160 170 Choices Description Check Primary DNS Server IP Address Check Internet Web Server Port Check Local IP Address Set Internet Connection Type Hear the IP address of the primary DNS (Domain Name Service) server. Hear the port number of the Internet Web server used for the Web-based Utility. Hear the local IP address of the System. Press 0 to use DHCP. Press 1 to use a static IP address. Press 2 to use PPPoE. Enter the IP address using numbers on the telephone keypad. Use the * (star) key when entering a decimal point. Enter the network or subnet mask using numbers on the telephone keypad. Use the * (star) key when entering a decimal point. Enter the IP address using numbers on the telephone keypad. Use the * (star) key when entering a decimal point. Select the type of Internet connection you are using. Refer to the documentation supplied by your Internet service provider. First, set the Internet Connection Type to static IP address; otherwise, you will hear, "Invalid Option," if you try to set the static IP address. First, set the Internet Connection Type to static IP address; otherwise, you will hear, "Invalid Option," if you try to set the network or subnet mask. 210 101 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. Set Static IP Address 111 Set Network (or Subnet) Mask 121 Set Gateway IP Address 131 First, set the Internet Connection Type to static IP address; otherwise, you will hear, "Invalid Option," if you try to set the gateway IP address. Chapter 5: Using the Interactive Voice Response Menu Using the Interactive Voice Response Menu 16 IP Telephony System Interactive Voice Response Menu Action Command (press these keys on the telephone) 161 Choices Description Set Primary DNS Server IP Address Enter the IP address using numbers on the telephone keypad. Use the * (star) key when entering a decimal point. Press 0 to select the router/NAT mode. Press 1 to select the bridge/switch mode. First, set the Internet Connection Type to static IP address; otherwise, you will hear, "Invalid Option," if you try to set the IP address of the primary DNS server. Use the router/NAT mode when the Internet phones are on the Local Area Network (LAN) side. Use the bridge/switch mode when the Internet phones are on the Wide Area Network (WAN) side. Set the Mode 201 Configure Auto-Attendant Messages Enable/Disable WAN Access to the Web-based Utility Manual Reboot 72255 Refer to the "Configuring the Auto-Attendant Messages" section at the end of this chapter. Press 1 to enable. Press 0 to disable. Use this setting to enable or disable WAN access to the Web-based Utility. (This Utility lets you configure the System.) After you hear, "Option successful," hang up the phone. The System will automatically reboot. Press 1 to confirm. Press * (star) to cancel. If necessary, enter the password. The System will request confirmation; enter 1 to confirm. You will hear, "Option successful." Hang up the phone. The System will reboot, and all settings will be reset to their factory default settings. 7932 732668 Factory Reset 73738 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." Chapter 5: Using the Interactive Voice Response Menu Using the Interactive Voice Response Menu 17 IP Telephony System Interactive Voice Response Menu Action Command (press these keys on the telephone) 79228 Choices Description Change Auto-Attendant Press 0 to use the auto-attendant based on day and time. Press 1 to use the Daytime Auto-Attendant. Press 2 to use the Nighttime Auto-Attendant. Press 3 to use the Weekend/Holiday Auto-Attendant. Use this setting to select the auto-attendant you want to use. You can have the auto-attendant change depending on the day and time, or you can use one auto-attendant for all days and hours. (Make sure the auto-attendant you select has been enabled through the Web-based Utility; otherwise, the auto-attendant feature will not work.) For more information, refer to "Chapter 6: Using the Web-based Utility." User Factory Reset 877778 Press 1 to confirm. Press * (star) to cancel. The System will request confirmation; enter 1 to confirm. You will hear, "Option successful." Hang up the phone. The System will reboot and all user-configurable settings will be reset to their factory default settings. Figure 5-1: Auto-Attendant Options Chapter 5: Using the Interactive Voice Response Menu Using the Interactive Voice Response Menu 18 IP Telephony System Entering a Password You may be prompted to enter a password when you want to reset the System to its factory default settings. To enter the password, use the phone's keypad, and follow the appropriate instructions. · To enter A, B, C, a, b, or c -- press 2. · To enter D, E, F, d, e, or f -- press 3. · To enter G, H, I, g, h, or i -- press 4. · To enter J, K, L, j, k, or l -- press 5. · To enter M, N, O, m, n, or o -- press 6. · To enter P, Q, R, S, o, q, r, or s -- press 7. · To enter T, U, V, t, u, or v -- press 8. · To enter W, X, Y, Z, w, x, y, or z -- press 9. · To enter all other characters, press 0. NOTE: These bulleted instructions only apply when you are entering a password. At all other times, pressing a number only selects a number, not a letter or punctuation mark. For example, to enter the password phone@321 by keypad, press these keys: 746630321. Then press the # (pound) key to indicate that you have finished entering the password. To cancel your entry and return to the main menu, press * (star). Configuring the Settings for Your Internet Phone Service If you want to change the settings for your Internet phone service, refer to the instructions provided by your ITSP and "Chapter 6: Using the Web-based Utility." Chapter 5: Using the Interactive Voice Response Menu Entering a Password 19 IP Telephony System Configuring the Auto-Attendant Messages The System provides a feature called the auto-attendant, which automatically answers incoming calls with greetings or directory messages. It can handle up to 10 incoming calls and uses the default user ID aa. Auto-Attendant Messages You can save up to 10 customized greetings. The first four have default messages, which can be changed through the Interactive Voice Response Menu. Prompt ID 1 2 3 4 Default Audio Message "If you know your party's extension, you may enter it now." "Your call has been forwarded." "Not a valid extension, please try again." "Goodbye." 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. Chapter 5: Using the Interactive Voice Response Menu Configuring the Auto-Attendant Messages 20 IP Telephony System 1 to Record a. If you entered 1, you will hear, "You may record your message after the tone. When finished, press #." b. After you record the message, you will hear, "To save, enter 1. To review, enter 2. To re-record, enter 3. To exit, enter *." c. Follow the instructions for the entry you have selected. If you entered 1, the new message will be saved. You will be returned to the menu described in step 5. If you entered 2, you will hear the message played. You will be returned to the menu described in step b. If you entered 3, you will be returned to the menu in step a. If you entered *, you will be returned to the menu in step 5. 2 to Review If you entered 2, you will hear the message played. You will be returned to the menu described in step 5. 3 to Delete a. If you entered 3, you will hear, "Enter 1 to confirm; enter * to exit." b. If you entered 1, the message will be erased. You will be returned to the menu described in step 5. If you entered *, you will be returned to the previous menu described in step 5. Figure 5-2: Auto-Attendant Message Options * 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. 21 Chapter 5: Using the Interactive Voice Response Menu Configuring the Auto-Attendant Messages IP Telephony System Chapter 6: Using the Web-based Utility Overview When you first install the System, Linksys strongly recommends that you use the Setup Wizard, which you can download from www.linksys.com. If you do not wish to run the Setup Wizard, you can use the Web-based Utility to configure the System. The System may have been pre-configured by your Internet Telephony Service Provider (ITSP), so you may not have to make any changes. If you do wish to make changes, follow the instructions in this chapter. The Web-based Utility offers two levels of access: user and admin (administrator). Your level of access depends on your service provider's policies. Also, access to some settings may be protected or blocked, so that service settings cannot be accidentally changed. For more information, contact your ITSP. This chapter will describe each web page of the Web-based Utility and each page's key functions. The Internet connection settings are configured on the Router - WAN Setup screen, while some of the most popular features: auto-attendant, music-on-hold, and call hunt are configured on the Voice - SIP screen. The Utility can be accessed via your web browser through use of a computer on your network. There are two main tabs: Router and Voice. Additional tabs will be available after you click one of the main tabs. NOTE: If you are not sure how to configure the settings, then keep the default settings. 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. Chapter 6: Using the Web-based Utility Overview 22 IP Telephony System · SIP. Configure service, music-on-hold, group paging, call hunt, and auto-attendant settings on this screen. In most cases, do not change service settings unless instructed to do so by your ITSP. · Provisioning. Use this screen to configure service provisioning settings. In most cases, you should not change these settings unless instructed to do by your ITSP. · Regional. Use this screen to configure call settings. In most cases, you should not change these settings unless instructed to do by your ITSP. · FXS 1/2. Use the appropriate screen to configure settings for each FXS (Phone) port on the System. · Line 1/2/3/4. Use the appropriate screen to configure settings for each external Internet phone line. How to Access the Web-based Utility To access the Web-based Utility of the System, launch Internet Explorer or Netscape Navigator on the administration computer connected to the System's Ethernet port. If the System uses its default address, then enter 192.168.0.1 in the Address field. If you have assigned a static IP address to the System, then enter in the Address field. Press the Enter key. Enter your user name and password. The default user name for administrative access is admin, and the default user name for user access is user. (These user names cannot be changed.) Then enter the password supplied by your ITSP. (By default, there is no password, so if you were not given a password, then leave this field blank.) To view the status information for the phones and their calls, click PBX Status. To switch to a different login, click User Login or Admin Login. Enter the appropriate login information. Two views of the Web-based Utility are available. Click basic to view basic settings, or click advanced to view advanced settings. When you have finished making changes on a screen, click the Submit All Changes button to save the changes, or click the Undo All Changes button to undo your changes. When changes are saved, the System may reboot. NOTE: If your ITSP supplied the System, then it may be pre-configured for you, and you do not need to change any settings. Refer to the instructions supplied by your ITSP for more information. The PBX Status Screen This screen shows status information for the phones and their calls. Registration This section shows the registration information for the phones. Registration. To remove a phone's registration, click its checkbox. Then click the Delete button. Chapter 6: Using the Web-based Utility How to Access the Web-based Utility 23 IP Telephony System Station. Shown here is the station name assigned to the phone. (This setting is configured through the phone.) User ID. Shown here is the extension number assigned to the phone. IP Address. Shown here is the local IP address of the phone. Reg Expires. This indicates the number of seconds left before the phone needs to re-register with the System. Parking Lot This section shows the calls that have been parked. Call park is a convenient feature that lets a call be put on hold and picked up from any extension number. Parking Lot. To remove a call from the Parking Lot, click its checkbox. Then click the Delete button. Caller ID. Shown here is the phone number of the caller. Parked By. Shown here is the extension number that parked the call. Figure 6-1: PBX Screen - Parking Lot 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. 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. Figure 6-2: PBX Screen - Inbound Call Figure 6-3: PBX Screen - Outbound Call Chapter 6: Using the Web-based Utility The PBX Status Screen 24 IP Telephony System The Router Tab The Router - Status Screen This screen displays product and system information. Product Information Product Name. Shown here is the model number of the System. Serial Number. Shown here is the serial number of the System. Software Version. Shown here is the version number of the System software. Hardware Version. Shown here is the version number of the System hardware. MAC Address. Shown here is the MAC address of the System. Client Certificate. Shown here is the status of the client certificate. It authenticates the System for use in the ITSP's network. Licenses. This indicates how many additional licenses you have acquired for the System. mac address: the unique address that a manufacturer assigns to each networking device. Figure 6-4: Router - Status Screen System Status Current Time. Displayed here is the current date and time of the System. Elapsed Time. Displayed here is the amount of time elapsed since the last reboot of the System. WAN Connection Type. Displayed here is the Internet connection type of the System. Current IP. Displayed here is the Internet IP address of the System. Host Name. Displayed here is the host name of the System. Domain. Displayed here is the domain name of the System. Current Netmask. Displayed here is the netmask or subnet mask of the System. Current Gateway. Displayed here is the IP address of the gateway. Primary DNS. Displayed here is the IP address of the primary DNS server. Chapter 6: Using the Web-based Utility The Router Tab 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. 25 IP Telephony System Secondary DNS. Displayed here is the IP address of the secondary DNS server. LAN IP Address. Displayed here is the local IP address of the System. Broadcast Pkts Sent. Displayed here is the number of broadcast packets sent. Broadcast Bytes Sent. Displayed here is the number of broadcast bytes sent. Broadcast Pkts Recv. Displayed here is the number of broadcast packets received and processed. Broadcast Bytes Recv. Displayed here is the number of broadcast bytes received and processed. Broadcast Pkts Dropped. Displayed here is the number of broadcast packets received but not processed. Broadcast Bytes Dropped. Displayed here is the number of broadcast bytes received but not processed. packet: a unit of data sent over a network. Chapter 6: Using the Web-based Utility The Router Tab 26 IP Telephony System The Router - WAN Setup Screen This screen lets you configure the Internet connection, MAC clone, remote management, QoS, VLAN, and optional settings. Information about your Internet connection type should be provided by your Internet Service Provider (ISP). If you do not have this information, contact your service provider. Internet Connection Settings Connection Type. Select the connection type you use: DHCP, Static IP, or PPPOE. If you already have a router for your network, select Static IP and assign an address that is appropriate for your network. (Refer to the router's documentation for more information about IP addressing.) Static IP Settings If you selected Static IP, complete the Static IP Settings section. Static IP. Enter the static or fixed IP address of the System (this should be provided by your ISP). NetMask. Enter the net or subnet mask of the System (this should be provided by your ISP). Gateway. Enter the IP address of the gateway (this should be provided by your ISP). Figure 6-5: Router - WAN Setup Screen 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 PPPOE Settings If you selected PPPOE, complete the PPPOE Settings section. PPPoE Login Name. Enter the name provided by your ISP. PPPOE Login Password. Enter the password provided by your ISP. PPPOE Service Name (optional). Enter the service name provided by your ISP. Optional Settings HostName. Enter the host name, if provided by your ISP. Domain. Enter the domain name, if provided by your ISP. Primary DNS. Enter the IP address of the primary DNS server. Secondary DNS (optional). Enter the IP address of the secondary DNS server. Chapter 6: Using the Web-based Utility The Router Tab 27 IP Telephony System DNS Server Order. Select the order in which the DNS servers should be used: Manual; Manual, DHCP; or DHCP, Manual. The default is Manual. DNS Query Mode. Select how the DNS servers should be queried: Parallel or Sequential. The default is Parallel. Primary NTP Server. Enter the IP address of the primary NTP server, which the System uses to keep the date and time current. Secondary NTP Server (optional). Enter the IP address of the secondary NTP server. MAC Clone Settings Enable MAC Clone Service. Select whether you want to clone a MAC address onto the System, yes or no. The default is no. Cloned MAC Address. Enter the MAC address you want to clone. Remote Management Enable WAN Web Server. This feature lets you enable or disable access to the Web-based Utility from the WAN side. Select yes or no from the drop-down menu. The default is no. WAN Web Server Port. Enter the port number used to access the Utility from the WAN side. The default is 80. QOS Settings QOS QDisc. QoS prioritizes voice communications when different types of traffic are competing for bandwidth. Select the method you want to use: NONE, CBQ, or TBF. The default is NONE. Maximum Uplink Speed. Enter the maximum upload speed of your Internet connection. The default is 128Kbps. VLAN (Virtual Local Area Network) Settings Enable VLAN. VLAN (802.1Q) settings let you use the System in a virtual LAN environment. Select yes or no from the drop-down menu. The default is no. VLAN ID. Enter the ID number used by the System. The default is 1. When you have finished making changes, click the Submit All Changes button to save the changes, or click the Undo All Changes button to undo your changes. Chapter 6: Using the Web-based Utility The Router Tab 28 IP Telephony System The Router - LAN Setup Screen This screen lets you configure the local network, dynamic DHCP, and static DHCP lease settings. Networking Service. Select the service you want to use, NAT or Bridge. The default is NAT. LAN Network Settings LAN IP Address. Enter the local IP address of the System. The default is 192.168.0.1. LAN Subnet Mask. Select the local subnet mask: 255.255.255.0, 255.255.255.128, 255.255.255.192, 255.255.255.224, 255.255.255.240, 255.255.255.248, or 255.255.255.252. The default is 255.255.255.0. Enable DHCP Server. To use the System as a router assigning IP addresses, select yes. Otherwise, select no. The default is yes. 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. Figure 6-6: Router - LAN Setup Screen 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. Chapter 6: Using the Web-based Utility The Router Tab 29 IP Telephony System 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. Figure 6-7: Router - Application Screen 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. 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. Chapter 6: Using the Web-based Utility The Router Tab 30 IP Telephony System The Voice Tab The Voice - Info Screen This screen shows voice-related settings for the System. Figure 6-8: Voice - Info Screen - Product Information Product Information Product Name. Shown here is the model number of the System. Serial Number. Shown here is the serial number of the System. Software Version. Shown here is the version number of the System software. Hardware Version. Shown here is the version number of the System hardware. MAC Address. Shown here is the MAC address of the System. Client Certificate. Shown here is the status of the client certificate, which indicates that the System has been authorized by your ITSP. Licenses. This indicates how many additional licenses you have acquired for the System. System Status Current Time. Displayed here is the current date and time of the System. Elapsed Time. Displayed here is the amount of time elapsed since the last reboot of the System. Figure 6-9: Voice - Info Screen - System Status FXS 1/2 Status The FXS 1 and FXS 2 ports are the Phone ports of the System. (You can connect analog phones or fax machines to both ports.) They have the same status information available. Hook State. Displayed here is the status of the phone's readiness. On indicates that the phone is ready for use, while Off indicates that the phone is in use. Message Waiting. This indicates whether you have new voicemail waiting. Call Back Active. This indicates whether a call back request is in progress. Last Called Number. Displayed here is the last number called. Chapter 6: Using the Web-based Utility The Voice Tab 31 IP Telephony System Last Caller Number. Displayed here is the number of the last caller. Calls 1 and 2 have the same status information available. Call 1/2 State. Displayed here is the status of the call. Call 1/2 Tone. Displayed here is the type of tone used by the call. Call 1/2 Encoder. Displayed here is the codec used for encoding. Call 1/2 Decoder. Displayed here is the codec used for decoding. Call 1/2 FAX. Displayed here is the status of the fax pass-through mode. Call 1/2 Type. Displayed here is the direction of the call. Call 1/2 Remote Hold. This indicates whether the far end has placed the call on hold. Call 1/2 Callback. This indicates whether the call was triggered by a call back request. Call 1/2 Peer Name. Displayed here is the name of the internal phone. Call 1/2 Peer Phone. Displayed here is the phone number of the internal phone. Call 1/2 Duration. Displayed here is the duration of the call. Call 1/2 Packets Sent. Displayed here is the number of packets sent. Call 1/2 Packets Recv. Displayed here is the number of packets received. Call 1/2 Bytes Sent. Displayed here is the number of bytes sent. Call 1/2 Bytes Recv. Displayed here is the number of bytes received. Call 1/2 Decode Latency. Displayed here is the number of milliseconds for decoder latency. Call 1/2 Jitter. Displayed here is the number of milliseconds for receiver jitter. Call 1/2 Round Trip Delay. Displayed here is the number of milliseconds for delay. Call 1/2 Packets Lost. Displayed here is the number of packets lost. Call 1/2 Packet Error. Displayed here is the number of invalid packets received. Chapter 6: Using the Web-based Utility The Voice Tab Figure 6-10: Voice - Info Screen - FXS Status 32 IP Telephony System Line 1/2/3/4 Status Lines 1, 2, 3, and 4 have the same status information available. Registration State. Shown here is the status of the line's registration with the ITSP. Last Registration At. Shown here are the last date and time the line was registered. Next Registration In. Shown here is the number of seconds until the next registration. Figure 6-11: Voice - Info Screen - Line Status Message Waiting. This indicates whether you have new voicemail waiting. Mapped SIP Port. Shown here is the port number of the mapped SIP port. Auto Attendant Prompt Status Prompt 1-4. The first four greetings or messages are defaults. If you change a default, then the screen will show the new prompt's duration in milliseconds. Prompt 5-10. For each prompt, the screen shows its duration in milliseconds. Space Remaining. Shown here is the number of milliseconds available. Current AA. Shown here is the auto-attendant in use. When you have finished making changes, click the Submit All Changes button to save the changes, or click the Undo All Changes button to undo your changes. Figure 6-12: Voice - Info Screen - Auto Attendant Prompt Status Chapter 6: Using the Web-based Utility The Voice Tab 33 IP Telephony System The Voice - System Screen This screen lets you configure system settings. IMPORTANT: In most cases, you should not change these settings unless instructed to do by your ITSP. System Configuration Figure 6-13: Voice - System Screen Restricted Access Domains. Enter the domain names permitted to access the System. Enable Web Admin Access. This setting lets you enable or disable local access to the Web-based Utility. Select yes or no from the drop-down menu. The default is yes. Admin Passwd. Enter the password for the administrator. (By default, there is no password.) User Password. Enter the password for the user. (By default, there is no password.) Miscellaneous Settings Syslog Server. Enter the IP address of the syslog server, which logs system information and critical events of the System. Debug Server. Enter the IP address of the debug server, which logs debug information of the System. Debug Level. This determines the level of debug information that will be generated. Select 0, 1, 2, or 3 from the drop-down menu. The higher the debug level, the more debug information will be generated. The default is 0, which indicates that no debug information will be generated. When you have finished making changes, click the Submit All Changes button to save the changes, or click the Undo All Changes button to undo your changes. Chapter 6: Using the Web-based Utility The Voice Tab 34 IP Telephony System The Voice - SIP Screen This screen lets you configure service, music-on-hold, group paging, call hunt, and auto-attendant settings. IMPORTANT: In most cases, you should not change the service settings unless instructed to do by your ITSP. Figure 6-14: Voice - SIP Screen - SIP Parameters SIP Parameters Max Forward. This is the SIP Max Forward value, which can range from 1 to 255. The default is 70. Max Redirection. This is the number of times an invite can be redirected to avoid an infinite loop. The default is 5. Max Auth. This is the maximum number of times (from 0 to 255) a request may be challenged. The default is 2. SIP User Agent Name. This is the User-Agent header used in outbound requests. The default is $VERSION. SIP Server Name. This is the Server header used in responses to inbound responses. The default is $VERSION. SIP Reg User Agent Name. This is the User-Agent name to be used in a REGISTER request. If this is not specified, then the SIP User Agent Name will also be used for the REGISTER request. SIP Accept Language. This is the Accept-Language header used by the System. There is no default (this indicates the System does not include this header). DTMF Relay MIME Type. This is the MIME Type used in a SIP INFO message to signal a DTMF event. The default is application/dtmf-relay. Hook Flash MIME Type. This is the MIME Type used in a SIP INFO message to signal a hook flash event. The default is application/hook-flash. Remove Last Reg. This feature lets you remove the last registration before registering a new one if the value is different. Select yes or no from the drop-down menu. The default is no. Use Compact Header. This feature lets you use compact SIP headers in outbound SIP messages. Select yes or no from the drop-down menu. The default is no. Escape Display Name. This feature lets you keep the Display Name private. Select yes if you want the System to enclose the string (configured in the Display Name) in a pair of double quotes for outbound SIP messages. Any Chapter 6: Using the Web-based Utility The Voice Tab 35 IP Telephony System occurrences of " or \ in the string will be escaped with \" and \\ inside the pair of double quotes. Otherwise, select no. The default is no. SIP Timer Values (sec) SIP T1. This is the RFC 3261 T1 value (RTT estimate), which can range from 0 to 64 seconds. The default is .5. SIP T2. This is the RFC 3261 T2 value (maximum retransmit interval for non-INVITE requests and INVITE responses), which can range from 0 to 64 seconds. The default is 4. Figure 6-15: Voice - SIP Screen - SIP Timer Values SIP T4. This is the RFC 3261 T4 value (maximum duration a message will remain in the network), which can range from 0 to 64 seconds. The default is 5. SIP Timer B. This is the INVITE time-out value, which can range from 0 to 64 seconds. The default is 32. SIP Timer F. This is the non-INVITE time-out value, which can range from 0 to 64 seconds. The default is 32. SIP Timer H. This is the INVITE final response, time-out value, which can range from 0 to 64 seconds. The default is 32. SIP Timer D. This is the ACK hang-around time, which can range from 0 to 64 seconds. The default is 32. SIP Timer J. This is the non-INVITE response, hang-around time, which can range from 0 to 64 seconds. The default is 32. INVITE Expires. This is the INVITE request Expires header value. If you enter 0, then the Expires header is not included in the request. The default is 240. ReINVITE Expires. This is the ReINVITE request Expires header value. If you enter 0, then the Expires header is not included in the request. The default is 30. Reg Min Expires. This is the minimum registration expiration time allowed from the proxy in the Expires header or as a Contact header parameter. If the proxy returns a value less than this setting, then the minimum value is used. The default is 1. Reg Max Expires. This is the maximum registration expiration time allowed from the proxy in the Min-Expires header. If the value is larger than this setting, then the maximum value is used. The default is 7200. Reg Retry Intvl. This is the interval to wait before the System retries registration after failing during the last registration. The default is 30. Chapter 6: Using the Web-based Utility The Voice Tab 36 IP Telephony System Reg Retry Long Intvl. When registration fails with a SIP response code that does not match, the System will wait for the specified length of time before retrying. If this interval is 0, then the System will stop trying. This value should be much larger than the Reg Retry Intvl value. The default is 1200. Response Status Code Handling SIT1-4 RSC. Enter the SIP response status code for the appropriate SIT Tone (SIT stands for Special Information Tone). For example, if you set the SIT1 RSC to 404, then when the user makes a call and a failure code of 404 is returned, the SIT1 tone is played. Try Backup RSC. This is the SIP response code that retries a backup server for the current request. Retry Reg RSC. This is the interval to wait before the System retries registration after failing during the last registration. Figure 6-16: Voice - SIP Screen - Response Status Code Handling RTP Parameters RTP Port Min. This is the minimum port number for RTP transmission and reception. The default is 16384. RTP Port Max. This is the maximum port number for RTP transmission and reception. The default is 16482. RTP Packet Size. This is the packet size in seconds, which can range from 0.01 to 0.16. Valid values must be a multiple of 0.01 seconds. The default is 0.030. Max RTP ICMP Err. This indicates that the RTP data stream has failed due to ICMP errors. The default is 0. RTCP Tx Interval. This is the interval for sending out RTCP sender reports on an active connection. It can range from 0 to 255 seconds. The default is 0. No UDP Checksum. Select yes if you want the System to calculate UDP header checksum for SIP messages. Otherwise, select no. The default is no. Stats in BYE. This sets whether the System will include the P-RTP-Stat header or response to a BYE message. The header contains RTP statistics of the current call. Select yes or no from the drop-down menu. The default is no. Figure 6-17: Voice - SIP Screen - RTP Parameters SDP Payload Types NSE Dynamic Payload. This is the NSE dynamic payload type. The default is 100. AVT Dynamic Payload. This is the AVT dynamic payload type. The default is 101. Figure 6-18: Voice - SIP Screen - SDP Payload Types Chapter 6: Using the Web-based Utility The Voice Tab 37 IP Telephony System INFOREQ Dynamic Payload. This is the INFOREQ dynamic payload type. There is no default. G726r16 Dynamic Payload. This is the G726-16 dynamic payload type. The default is 98. G726r24 Dynamic Payload. This is the G726-24 dynamic payload type. The default is 97. G726r40 Dynamic Payload. This is the G726-40 dynamic payload type. The default is 96. G729b Dynamic Payload. This is the G729b dynamic payload type. The default is 99. NSE Codec Name. This is the NSE codec name used in SDP. The default is NSE. AVT Codec Name. This is the AVT codec name used in SDP. The default is telephone-event. G711u Codec Name. This is the G711u codec name used in SDP. The default is PCMU. G711a Codec Name. This is the G711a codec name used in SDP. The default is PCMA. G726r16 Codec Name. This is the G726-16 codec name used in SDP. The default is G726-16. G726r24 Codec Name. This is the G726-24 codec name used in SDP. The default is G726-24. G726r32 Codec Name. This is the G726-32 codec name used in SDP. The is G726-32. G726r40 Codec Name. This is the G726-40 codec name used in SDP. The default is G726-40. G729a Codec Name. This is the G729a codec name used in SDP. The default is G729a. G729b Codec Name. This is the G729b codec name used in SDP. The default is G729ab. G723 Codec Name. This is the G723 codec name used in SDP. The default is G723. NAT Support Parameters Handle VIA received. If you select yes, the System will process the received parameter in the VIA header (this is inserted by the server in a response to any one of its requests). If you select no, the parameter will be ignored. Select yes or no from the drop-down menu. The default is no. Handle VIA rport. If you select yes, the System will process the rport parameter in the VIA header (this is inserted by the server in a response to any one of its requests). If you select no, the parameter will be ignored. Select yes or no from the drop-down menu. The default is no. Figure 6-19: Voice - SIP Screen - NAT Support Parameters Chapter 6: Using the Web-based Utility The Voice Tab 38 IP Telephony System Insert VIA received. This lets you insert the received parameter into the VIA header of SIP responses if the received-from IP and VIA sent-by IP values differ. Select yes or no from the drop-down menu. The default is no. Insert VIA rport. This feature lets you insert the rport parameter into the VIA header of SIP responses if the received-from port and VIA sent-by port numbers differ. Select yes or no from the drop-down menu. The default is no. Substitute VIA Addr. This feature lets you use NAT-mapped IP:port values in the VIA header. Select yes or no from the drop-down menu. The default is no. Send Resp To Src Port. This feature lets you send responses to the request source port instead of the VIA sentby port. Select yes or no from the drop-down menu. The default is no. STUN Enable. This feature lets you use STUN to discover NAT mapping. Select yes or no from the drop-down menu. The default is no. STUN Test Enable. If the STUN Enable feature is enabled and a valid STUN server is available, then the System can perform a NAT type discovery operation when it powers on. It will contact the configured STUN server, and the result of the discovery will be reported in a Warning header in all subsequent REGISTER requests. If the System detects symmetric NAT or a symmetric firewall, NAT mapping will be disabled. The STUN Test Enable feature lets you use the STUN test. Select yes or no from the drop-down menu. The default is no. STUN Server. Enter the IP address of the STUN server to contact for NAT mapping discovery. EXT IP. Enter the external IP address to substitute for the actual IP address of the System in all outgoing SIP messages. If 0.0.0.0 is specified, then no IP address substitution will be performed. EXT RTP Port Min. This is the external port mapping number of the RTP Port Min. number. If this value is not zero, then the RTP port number in all outgoing SIP messages will be substituted for the corresponding port value in the external RTP port range. NAT Keep Alive Intvl. This is the interval between NAT-mapping, keep alive messages. The default is 15. PBX Parameters Proxy Network Interface. This tells the System how the clients (usually phones) are connected. Select LAN or WAN. The default is WAN. Proxy Listen Port. This is the port used by the System when it listens for client messages at the selected interface. The default is 6060. Chapter 6: Using the Web-based Utility The Voice Tab Figure 6-20: Voice - SIP Screen - PBX Parameters 39 IP Telephony System Multicast Address. This is the IP address (and port number) used by the System to send control messages to all clients at the same time. It must be a multicast address and must contain a port number. The default is 224.168.168.168:6061. Group Page Address. This is the IP address (and port number) used by the System to tell clients to send and receive group page RTP packets. It must be a multicast address and must contain a port number. The default is 244.168.168.168:34567. Max Expires. This sets the maximum allowed Registration Expires value (in seconds) for clients. The default is 3600. Force Media Proxy. This feature forces external clients to use the System's media proxy when exchanging RTP traffic with external peers. Select yes or no from the drop-down menu. The default is no. Proxy Debug Option. SIP messages are received at or sent from the proxy listen port. This feature controls which SIP messages to log. Select none for no logging. Select 1-line to log the start-line only for all messages. Select 1-line excl. OPT to log the start-line only for all messages except OPTIONS requests/responses. Select 1-line excl. NTFY to log the start-line only for all messages except NOTIFY requests/responses. Select 1-line excl. REG to log the start-line only for all messages except REGISTER requests/responses. Select 1-line excl. OPT|NTFY|REG to log the start-line only for all messages except OPTIONS, NOTIFY, and REGISTER requests/responses. Select full to log all SIP messages in full text. Select full excl. OPT to log all SIP messages in full text except OPTIONS requests/responses. Select full excl. NTFY to log all SIP messages in full text except NOTIFY requests/responses. Select full excl. REG to log all SIP messages in full text except REGISTER requests/responses. Select full excl. OPT|NTFY|REG to log all SIP messages in full text except for OPTIONS, NOTIFY, and REGISTER requests/responses. The default is full. Call Routing Rule. This is a special dial plan that determines which line can be used for an external, outbound call request from a phone based solely on the target public number. When you create this rule, follow this format: (rule|rule|rule|...|rule) The most specific rules should be placed first. Each rule should be in this format: <:Lx>pattern L indicates Line (phone line). The variable x is 1, 2, 3, or 4 depending on which line you want to specify. The word pattern indicates any digit pattern (see the Dial Plan setting for more information). Chapter 6: Using the Web-based Utility The Voice Tab 40 IP Telephony System The default is (9xx.); this indicates that any of the four lines can be used for any target number starting with 9. For example, with this dial plan, the caller dials 9 before entering the external phone number. Internal Music URL. Enter the Uniform Resource Locator (URL), also known as a web address, to download a music file for the music-on-hold and call park features. This is its format: [tftp://]server_IP_address[:port]/path. TFTP is the only protocol supported for music download. The default port is 69. Saving a new URL will reboot the System. After its reboot, the System will download the file and store the samples in flash memory. The music samples are encoded in G711u format at 8000 samples/second. This file should not contain any extra header information, and its maximum length is 65.536 seconds (524,288 bytes). For more information, refer to "Appendix D: New Music for the Music-on-Hold Feature." Internal Music Script. This script tells the System how to play the downloaded music file. This is its format: [section[,(section[,...]]] Each section should be in this format: [n (start/end[/pause])] [pause2] The variable n is the number of times you want a section to repeat before moving to the next section. The start/end is the starting and 1+ending sample for the section. Note that the samples are numbered from 0 to total-length - 1. You may enter -1 or a very large number if the end of the file should be the ending sample. The default start value is 0, and the default end value is the end of the file. The variable pause is the number of samples to pause after the ending sample has been played. The default is 0. The variable pause2 is the additional number of samples to pause after the entire n repetitions of the section have been played. The default is 0. A maximum of 16 sections can be specified. Samples should be encoded in G711u format at 8000 samples/second. After all sections are played, they are replayed, starting with the first section. For example, the default Internal Music Script setting is 2(0/230954),2(230954/444720),(0/230954)40000. The first section is 2(0/230954); samples 0 through 230954 will be played twice. The second section is 2(230954/444720); samples 230954 through 444720 will be played twice. The third section is (0/230954); samples 0 through 230954 will be played once. The last section is 40000. The ending pause will last for 40,000 samples. Each sample lasts 1/8000 of a second, so 40,000 samples equals 5 seconds. When this pause is over, the sections are replayed. Internal MOH Refresh Intvl. The System can refresh an internal music session periodically. The default is 0, which disables the refresh function. Chapter 6: Using the Web-based Utility The Voice Tab 41 IP Telephony System Call Park MOH Server. Enter the name or IP address of the music-on-hold server that should be used to handle a parked call. If you do not have a music-on-hold server for the call park feature, then keep the default, imusic, and the parked caller will hear the internal music file. Otherwise, if this setting is not specified, the parked caller will hear silence. Call Park DLG Refresh Intvl. The System can refresh a call park session periodically. The default is 0, which disables the refresh function. Default Group Line. This is the default group of lines, 1,2,3,4. Group 1-4 User ID. A group designates specific phones that should be paged as a group, use the same phone lines, and receive the same type of calls. For example, sales calls should go to the sales group. You can designate up to four groups. For each group, enter a comma-separated list of User IDs, each representing a different client. For example, if the sales group is Group 1, then enter the sales extensions: 501,502,503 in the Group 1 User ID field. A client can belong to more than one group. If a client does not belong to any group, then the client belongs to the default group assigned to the Default Group Line. Each User ID pattern can use * and ? wildcards as well as %xx escaped characters (refer to "Appendix C: Dial Plan and Auto-Attendant Scripting for Advanced Users" for more details). The default is a blank field, which means that all clients belong to the default group. Group 1-4 Line. For each group, enter a comma-separated list of phone lines the clients can use (this list determines the order in which the lines will be used). The System will make external calls for clients using the phone lines listed here. For example, for a group whose setting is 1,3, then System will use Line 1. If that is not successful, then it will use Line 3. Hunt Groups. This defines one or more hunt groups that can be called directly by any client like a regular extension. The syntax is the same as the syntax for the Contact List. Note that a member of one group can also be the extension of another group (i.e., one level of recursion is allowed). SIP DIDN Field. This determines which field is used to indicate the Direct Inward Dialing (DID) number for an incoming INVITE to a line interface. Select TO UserID to use the User-ID field of the TO header, or select TO Param to use a parameter in the TO header with the name specified in the SIP DIDN PARAM Name. The default is TO UserID. SIP DIDN Param Name. This indicates the DID number in an incoming INVITE message. The default is didn. Auto Attendant Parameters AA Dial Plan 1. This is used to define the first dial rule in the auto-attendant. The default is (10x|xxx.). Refer to "Appendix C: Dial Plan and Auto-Attendant Scripting for Advanced Users" for more details. AA Dial Plan 2. This is used to define the second dial rule in the auto-attendant. The default is (<:10>x|xxx.). Chapter 6: Using the Web-based Utility The Voice Tab Figure 6-21: Voice - SIP Screen - Auto Attendant Parameters 42 IP Telephony System AA script 1-3. These are used to define the three auto-attendant scripts. Refer to "Appendix C: Dial Plan and Auto-Attendant Scripting for Advanced Users" for more details. DayTime AA. To enable the daytime auto-attendant, select yes. Otherwise, select no. The default is yes. DayTime. Enter the daytime hours for the daytime auto-attendant in 24-hour format. Enter the start and end times in this format: start=hh:mm:ss;end=hh:mm:ss (hh for hours, mm for minutes, and ss for seconds) For example, start=9:0:0;end=17:0:0 means the start time is 9 AM and the end time is 5 PM. The other hours (5 PM to 9 AM) are considered nighttime hours. If you do not enter start and end times, then the whole day (24 hours) is considered as daytime, so the nighttime auto-attendant will not be used, even if it is enabled. DayTime AA Script. Select the daytime auto-attendant script that you want to use, 1, 2, or 3. The default is 1. DayTime Answer Delay. Select the number of seconds you want the daytime auto-attendant to wait before answering. The default is 12 seconds. NightTime AA. To enable the nighttime auto-attendant, select yes. Otherwise, select no. The default is no. NightTime AA Script. Select the nighttime auto-attendant script that you want to use, 1, 2, or 3. The default is 1. NightTime Answer Delay. Select the number of seconds you want the nighttime auto-attendant to wait before answering. The default is 0 seconds. Weekend/Holiday AA. To enable this auto-attendant, select yes. Otherwise, select no. The default is no. Weekends/Holidays. When the weekend/holiday auto-attendant is enabled, you can use this setting to specify the weekends and holidays. Up to four weekend days can be defined. Use this format: [wk=n1[,ni];][hd=mm/dd/yyyy|mm/dd/yyyy-mm/dd/yyyy[,mm/dd/yyyy|mm/dd/yyyy-mm/dd/yyyy];] (wk for weekend, which can be 1 for Monday to 7 for Sunday) (hd for holiday, which does not have to include the year) For example, wk=6,7;hd=1/1,2/21/2006,5/30/2006,12/19/2006-12/30/2006 means that Saturdays and Sundays are the weekends. Holidays are January 1-2, 2006; May 30, 2006; and December 19-30, 2006. Chapter 6: Using the Web-based Utility The Voice Tab 43