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Linksys PAP2T Product Brochure
Linksys PAP2T Quick Installation
Linksys PAP2 Product Brochure
The Linksys Phone Adapter enables high-quality feature-rich telephone service through your cable or DSL Internet connection. Just plug it into your home router or gateway and use the two standard telephone jacks to connect your existing phones or fax machines. Each phone jack operates independently, with separate phone service and phone numbers - like having two phone lines. With an appropriate Internet telephone service provider, you'll get clear telephone reception and reliable fax connect... Read more
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User reviews and opinions
|firstvoder||10:48pm on Wednesday, May 19th, 2010|
|Wrong product delivered the item description is "Cisco PAP2 Phone Adapter for Vonage Internet Phone Service".|
|Jankebek||2:08am on Monday, May 10th, 2010|
|excellent product and a wonderfull seller this is a very good product from cisco i try to find it first because of the quality and indurance the selle... Very easy to use. We have been using this adapter for 2 years. We have never had any problems with the adapter itself.|
Comments posted on www.ps2netdrivers.net are solely the views and opinions of the people posting them and do not necessarily reflect the views or opinions of us.
Internet Phone Adapter
with 2 Ports for Voice-over-IP
Internet Phone Adapter with 2 Ports for Voice-over-IP
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
Your guide to the Internet Phone Adapter has been designed to make understanding networking with the Internet Phone Adapter 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 Phone Adapter. This exclamation point means there is a caution or warning and is something that could damage your property or the Phone Adapter. This question mark provides you with a reminder about something you might need to do while using the Phone Adapter. 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.
Table of Contents
Chapter 1: Introduction
... Welcome. Whats in this Guide?. Chapter 2: Getting to Know the Phone Adapter. The Back Panel. The Front Panel. Chapter 3: Connecting the Phone Adapter. Overview. Connecting the Phone Adapter. 5
Chapter 4: Using the Phone Adapters Interactive Voice Response Menu.
7 Overview. 7 Accessing the Interactive Voice Response Menu. 7 Using the Interactive Voice Response Menu. 8 Entering a Password. 12 Configuring the Settings for Your Internet Phone Service. 12 Appendix A: Troubleshooting. 13 Common Problems and Solutions. 13 Frequently Asked Questions. 16 Appendix B: Glossary. 17
Appendix C: Specifications. Appendix D: Warranty Information. Appendix E: Regulatory Information. Appendix F: Contact Information.
33 Internet Phone Service Provider. 33 Linksys. 33
Analog Telephone Adapter with 2 FXS Ports
List of Figures
Figure 2-1: Back Panel. Figure 2-2: Front Panel. Figure 3-1: Connection Diagram of a Typical Setup. Figure 3-2: Connect the RJ-11 Telephone Cable. Figure 3-3: Connect the Ethernet Network Cable. Figure 3-4: Connect the Power Adapter. 6 6
Thank you for choosing the Linksys Internet Phone Adapter with 2 Ports for Voice-over-IP. This Phone Adapter will allow you to make phone or fax calls using your broadband connection. How does the Phone Adapter do this? Plug the Phone Adapter into your network router or gateway, then connect your phones or fax machines to the Phone Adapter. Each of the Phone Adapters two telephone jacks operate independently, with separate Internet phone services and phone numbers, so you can have up to two Internet phone lines delivered through your cable or DSL Internet connection. What does all of this mean? The Phone Adapter allows your phones or fax machines to take advantage of and share your high-speed Internet connection. You can even make phone calls while youre surfing the Internet (Internet phone services required). Refer to the instructions in the Quick Installation or this User Guide to help you connect the Phone Adapter after you have set up an account with your Internet phone service provider. The instructions in this User Guide will help you get the most out of your Phone Adapter.
NOTE: For information about your Internet phone service, refer to the materials provided by your Internet phone service provider.
Chapter 1: Introduction Welcome
Whats in this Guide?
This User Guide covers the steps for installing the Phone Adapter. Chapter 1: Introduction Describes the Phone Adapters applications and the contents of this User Guide. Chapter 2: Getting to Know the Phone Adapter Describes the physical features of the Phone Adapter. Chapter 3: Connecting the Phone Adapter Explains how to connect the Phone Adapter to your network and phones (or fax machines). Chapter 4: Using the Phone Adapters Interactive Voice Response Menu Explains how to configure the Phone Adapters network settings using the Interactive Voice Response Menu. Appendix A: Troubleshooting Describes potential problems and solutions regarding use of the Phone Adapter. Appendix B: Glossary Gives brief descriptions of terms frequently used in networking. Appendix C: Specifications Provides technical specifications for the Phone Adapter. Appendix D: Warranty Information Provides warranty information for the Phone Adapter. Appendix E: Regulatory Information Provides regulatory information regarding the Phone Adapter. Appendix F: Contact Information Provides contact information for a variety of Linksys resources, including Technical Support.
Chapter 1: Introduction Whats in this Guide?
Chapter 2: Getting to Know the Phone Adapter
The Back Panel
The Phone Adapters ports are located on the back panel.
Figure 2-1: Back Panel
The PHONE 1 port (for your primary Internet phone line) allows you to connect your telephone to the Phone Adapter using an RJ-11 telephone cable. If you have a second Internet phone service account, use the PHONE 2 port to connect a second telephone (or fax machine) to the Phone Adapter using an RJ-11 telephone cable. The INTERNET port allows you to connect the Phone Adapter to your router or gateway using a Category 5 (or better) Ethernet network cable. The POWER port is where you connect the included power adapter.
PHONE 2 Port
Chapter 2: Getting to Know the Phone Adapter The Back Panel
The Front Panel
The Phone Adapters LEDs, which inform you about network activities, are located on the front panel.
Figure 2-2: Front Panel
Green. The Power LED lights up when the Phone Adapter is powered on and connected to the Internet. It flashes when the Phone Adapter is not connected to the Internet. The LED also flashes when the Phone Adapter is booting up or upgrading its firmware. Green. The INTERNET LED lights up when there is an active connection through the port. If the LED is flashing, then there is traffic moving through that port. Green. The PHONE LED is solidly lit when a telephone or fax machine has an active or registered connection to your Internet phone service provider through the corresponding port. It flashes when the phone is being used or is off the hook.
Chapter 3: Connecting the Phone Adapter
This chapter explains how to connect the Phone Adapter to your network and telephones or fax machines. Shown below is a connection diagram displaying a typical setup.
Figure 3-1: Connection Diagram of a Typical Setup
Connecting the Phone Adapter
If you already have an account set up with your Internet phone service provider, then proceed to step 1. If you do not have an account set up with your Internet phone service provider, then go to your providers website and sign up for service. After you have set up an account, proceed to step 1. 1. Using a telephone cable, connect your telephone to the PHONE 1 port of the Phone Adapter. 2. If you have a second Internet phone service account, connect another telephone or a fax machine to the PHONE 2 port of the Phone Adapter.
IMPORTANT: The Phone Adapter includes a ringer (ring signal generator), which is a source of hazardous voltage. When the ringer is activated by an incoming call, do not touch the Phone port wires, the wires of a cable connected to either of the Phone ports, or the internal circuitry of the Phone Adapter. IMPORTANT: Do not connect either of the Phone ports to a telephone wall jack. Make sure you only connect a telephone or fax machine to either of the Phone ports. Otherwise, the Phone Adapter or the telephone wiring in your home or office may be damaged.
Figure 3-2: Connect the RJ-11 Telephone Cable
Chapter 3: Connecting the Phone Adapter Overview
3. Connect the included Ethernet network cable to the INTERNET port of the Phone Adapter. Connect the other end to one of the Ethernet ports on your router or gateway. 4. Connect the included power adapter to the POWER port on the back panel of the Phone Adapter. Connect the other end to a standard electrical outlet. 5. The Status and Act LEDs will be solidly lit when the Phone Adapter is ready for use. 6. Follow the instructions provided by your Internet phone service provider. If you need to manually configure the Phone Adapters network settings, you can use the telephone connected to the Phone Adapter. See Chapter 4: Using the Phone Adapters Interactive Voice Response Menu for instructions. The installation of the Phone Adapter is complete. Now you can pick up your phone and make calls. Figure 3-3: Connect the Ethernet Network Cable
Figure 3-4: Connect the Power Adapter
Chapter 3: Connecting the Phone Adapter Connecting the Phone Adapter
Chapter 4: Using the Phone Adapters Interactive Voice Response Menu
You may need to manually configure the Phone Adapter by entering the settings provided by your Internet phone service provider. This chapter explains how to use the Interactive Voice Response Menu to configure the Phone Adapters network settings. You will use the phones keypad to enter your commands and select choices, and the Phone Adapter will respond with voice responses. NOTE: If your Internet phone service provider sent you the Phone Adapter, then it may be pre-configured for you, and you may not need to change any settings. Refer to the instructions provided by your service provider for more information.
Accessing the Interactive Voice Response Menu
1. Use a telephone connected to the PHONE 1 or PHONE 2 port of the Phone Adapter. 2. Press the * (star) key four times. 3. Wait until you hear Configuration menu. Please enter option followed by the # (pound) key or hang up to exit. 4. Refer to the table on page 8 for a list of actions, commands, menu choices, and descriptions. After you select an option, press the # (pound) key. To exit the menu, hang up the telephone.
Chapter 4: Using the Phone Adapters Interactive Voice Response Menu Overview
Using the Interactive Voice Response Menu
When entering values, such as an IP address, you may exit without saving your changes by press the * (star) key twice within half a second. Otherwise, the * (star) key 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 the 1 key. To review the new setting, press the 2 key. To re-enter the new setting, press the 3 key. To cancel your entry and return to the main menu, press the * (star) key. For example, to enter the IP address 188.8.131.52 using your telephones keypad, press: 191*168*1*105. When you are finished, press the # (pound) key, then press the 1 key to save the IP address. To cancel your entry and return to the main menu, press the * (star) key. If the menu is inactive for more than one minute, the Phone Adapter will automatically exit the Interactive Voice Response Menu. To re-enter the menu, press the * (star) key four times. The settings you have saved will take effect when you hang up the telephone. The Phone Adapter may reboot at this time.
Interactive Voice Response Menu
Action Command (press these keys on the telephone)
Enter Interactive Voice Response Menu
Use this command to enter the Interactive Voice Response Menu. Do not press any other keys until you hear, Configuration menu. Please enter option followed by the # (pound) key or hang up to exit. Use this command to find out if DHCP has been enabled or disabled. If it is enabled, the Phone Adapter will be assigned an IP address automatically by your network router or gateway. If it is disabled, then the Phone Adapter will need to use a static IP address.
Chapter 4: Using the Phone Adapters Interactive Voice Response Menu Using the Interactive Voice Response Menu
Enter 1 to enable Enter 0 to disable
Enable or disable the Phone Adapters DHCP feature. If your network router assigns IP addresses, enter 1, otherwise, enter 0. Use this command to check the current IP address of the Phone Adapter. To set a static IP address, the DHCP feature must be disabled. If the DHCP feature is enabled, you will hear, Invalid Option, if you try to set a static IP address. Use this command to check the current network mask of the Phone Adapter. To set the network mask, the DHCP feature must be disabled. If the DHCP feature is enabled, you will hear, Invalid Option, if you try to set the network mask. Use this command to check the current gateway IP address of the Phone Adapter. To set a static gateway IP address, the DHCP feature must be disabled. If the DHCP feature is enabled, you will hear, Invalid Option, if you try to set a static gateway IP address.
Check IP Address
Set Static IP Address
Enter the IP address using the numbers on your telephone keypad. Use the * (star) key to enter a decimal point.
Check Network Mask (or Subnet Mask) Set Network Mask (or Subnet Mask)
Enter the network mask using numbers on your telephone keypad. Use the * (star) key to enter a decimal point.
Check Static Gateway IP Address Set Static Gateway IP Address
Enter the IP address using numbers on your telephone keypad. Use the * (star) key to enter a decimal point.
Check MAC Address
Use this command to check the current MAC address of the Phone Adapter in hexadecimal string format. Use this command to check the current version number of the firmware currently installed on the Phone Adapter. This feature is password protected by your Internet phone service provider. (The Web-based Utility is disabled by default because you will not need to use it unless instructed to do so by your service provider.) If you have been provided a password, see Entering a Password.
Check Firmware Version
Enable/Disable Webbased Utility of Phone Adapter
After you hear, Option successful, hang up the phone. The Phone Adapter will automatically reboot.
Enter 1 to confirm Enter * (star) to cancel
This feature is password protected by your Internet phone service provider. (You will not need to reset the Phone Adapter to its factory default settings unless instructed to do so by your service provider.) If you have been provided a password, see Entering a Password. After you have entered your password, the Phone Adapter will request confirmation. After confirmation, you will hear, Option successful. Hang up the phone. The Phone Adapter will reboot and all settings will be reset to their factory default settings.
Entering a Password
You will be prompted to enter a password if you want to enable the Web-based Utility or reset the Phone Adapter to its factory default settings. To enter a password, use the phones keypad, and follow the appropriate instructions. To enter A, B, C, a, b, or c press the 2 key. To enter D, E, F, d, e, or f press the 3 key. To enter G, H, I, g, h, or i press the 4 key. To enter J, K, L, j, k, or l press the 5 key. To enter M, N, O, m, n, or o press the 6 key. To enter P, Q, R, S, o, q, r, or s press the 7 key. To enter T, U, V, t, u, or v press the 8 key. To enter W, X, Y, Z, w, x, y, or z press the 9 key. To enter all other characters, press the 0 key. NOTE: These bulleted instructions above only apply when you are entering a password. Pressing a number at all other times selects only a number, not a letter or punctuation mark. For example, to enter the password phone@321 by keypad, press: 746630321. When you are finished, press the # (pound) key. To cancel your entry and return to the main menu, press the * (star) key.
Configuring the Settings for Your Internet Phone Service
If you want to change the settings for your Internet phone service, visit your Internet phone service providers website and make configuration changes online. For more information, refer to the instructions provided by your Internet phone service provider.
Chapter 4: Using the Phone Adapters Interactive Voice Response Menu Entering a Password
Appendix A: Troubleshooting
This appendix provides solutions to problems you may experience when installing and using the Phone Adapter. Read the descriptions below to solve your problems. If you can't find an answer here, check your Internet phone service providers website.
Common Problems and Solutions
1. I dont hear a dial tone. Go through this checklist until your problem is solved: Make sure the telephone is plugged into the correct port, PHONE 1 or PHONE 2. (You should use the Phone 2 port only if you have more than one Internet telephone service account.) Make sure the green INTERNET LED on the front panel of the Phone Adapter is lit. If it is not lit, check your router and network connection to the Internet. Make sure the green PHONE 1 LED on the front panel of the Phone Adapter is lit, then pick up the telephone; you should now have a dial tone. Follow these instructions to reboot your cable or DSL modem, Phone Adapter, and router: 1. Power off your cable or DSL modem by unplugging its power adapter. 2. Power off the router by unplugging its power adapter. 3. Power off the Phone Adapter by unplugging its power adapter. 4. Wait two minutes, then power on your cable or DSL modem by plugging its power adapter into an electrical outlet. 5. Wait two minutes, then power on the router by plugging its power adapter into an electrical outlet. 6. Wait five minutes, then power on the Phone Adapter by plugging its power adapter into an electrical outlet. 7. Reboot one of your networked computers, and check to see if you have an active Internet connection. 8. Pick up the telephone. You should now have a dial tone. Set up port forwarding on your router. You must specify that ports 5060-5061 (UDP), 53-53 (UDP), 69-69 (UDP), and 10000-20000 (UDP) are forwarded to the IP address of the Phone Adapter. Before you set up port forwarding on your router, change two of the Phone Adapters settings using its Interactive Voice Response Menu. First, disable its DHCP feature, then assign a static IP address to the Phone Adapter. See Chapter 4: Using the Interactive Voice Response Menu for instructions.
Appendix A: Troubleshooting Common Problems and Solutions
2. When I make a telephone call, the call doesnt go through; instead, I hear a fast busy signal. Go through this checklist until your problem is solved: Follow these instructions to reboot your cable or DSL modem, Phone Adapter, and router: 1. Power off your cable or DSL modem by unplugging its power adapter. 2. Power off the router by unplugging its power adapter. 3. Power off the Phone Adapter by unplugging its power adapter. 4. Wait two minutes, then power on your cable or DSL modem by plugging its power adapter into an electrical outlet. 5. Wait two minutes, then power on the router by plugging its power adapter into an electrical outlet. 6. Wait five minutes, then power on the Phone Adapter by plugging its power adapter into an electrical outlet. 7. Reboot one of your networked computers, and check to see if you have an active Internet connection. 8. Pick up the telephone. You should now have a dial tone. Set up port forwarding on your router. You must specify that ports 5060-5061 (UDP), 53-53 (UDP), 69-69 (UDP), and 10000-20000 (UDP) are forwarded to the IP address of the Phone Adapter. Before you set up port forwarding on your router, change two of the Phone Adapters settings using its Interactive Voice Response Menu. First, disable its DHCP feature, then assign a static IP address to the Phone Adapter. See Chapter 4: Using the Interactive Voice Response Menu for instructions.
3. When Im on a telephone call, words are dropped intermittently. Go through this checklist until your problem is solved: If youre using a wireless router and a cordless phone, they may be using the same frequency and may interfere with each other. Move the cordless phone farther away from the router. There may be heavy network activity, particularly if you are running a server or using a file sharing program. Try to limit network or Internet activity during Internet telephone calls. For example, if you are running a file sharing program, file may be uploaded in the background even though you are not downloading any file, so make sure you exit the program before making Internet phone calls. There may not be enough bandwidth available for your Internet phone call. You may want to test your bandwidth using one of the bandwidth tests available online. If necessary, access your Internet phone service account and reduce the bandwidth requirements for your service. For more information, refer to your service providers website.
4. My telephone does not ring, and my calls automatically go to voicemail. Go through this checklist until your problem is solved: Make sure the telephone is plugged into the correct port, PHONE 1 or PHONE 2. (You should use the Phone 2 port only if you have more than one Internet telephone service account.) Make sure the ringer volume on your telephone is set to an audible level. Follow these instructions to reboot your cable or DSL modem, Phone Adapter, and router: 1. Power off your cable or DSL modem by unplugging its power adapter. 2. Power off the router by unplugging its power adapter. 3. Power off the Phone Adapter by unplugging its power adapter. 4. Wait two minutes, then power on your cable or DSL modem by plugging its power adapter into an electrical outlet. 5. Wait two minutes, then power on the router by plugging its power adapter into an electrical outlet. 6. Wait five minutes, then power on the Phone Adapter by plugging its power adapter into an electrical outlet. 7. Reboot one of your networked computers, and check to see if you have an active Internet connection. Set up port forwarding on your router. You must specify that ports 5060-5061 (UDP), 53-53 (UDP), 69-69 (UDP), and 10000-20000 (UDP) are forwarded to the IP address of the Phone Adapter. Before you set up port forwarding on your router, change two of the Phone Adapters settings using its Interactive Voice Response Menu. First, disable its DHCP feature, then assign a static IP address to the Phone Adapter. See Chapter 4: Using the Interactive Voice Response Menu for instructions.
5. The INTERNET LED does not light up. Go through this checklist until your problem is solved: Check the cable connecting the Phone Adapter to your router. Make sure your network has an active Internet connection. If it does not, follow these instructions to reboot your cable or DSL modem, Phone Adapter, and router: 1. Power off your cable or DSL modem by unplugging its power adapter. 2. Power off the router by unplugging its power adapter. 3. Power off the Phone Adapter by unplugging its power adapter. 4. Wait two minutes, then power on your cable or DSL modem by plugging its power adapter into an electrical outlet. 5. Wait two minutes, then power on the router by plugging its power adapter into an electrical outlet. 6. Wait five minutes, then power on the Phone Adapter by plugging its power adapter into an electrical outlet. 7. Reboot one of your networked computers, and check to see if you have an active Internet connection.
Frequently Asked Questions
How do I make a phone call? Pick up your phone and dial a phone number. Some Internet phone service providers may require specific dialing instructions. Please check with your Intenet phone service provider for details. Can I make calls if my Internet connection is down? No. Your high-speed Internet connection must be active when you make Internet phone or fax calls. Can I make calls while Im browsing the Internet? Yes. You can make Internet phone or fax calls while browsing the Internet, however, upstream data traffic passing through your Internet connection may affect the quality of your telephone call.
Appendix A: Troubleshooting Frequently Asked Questions
Appendix B: Glossary
This glossary contains some basic networking terms you may come across when using this product. For more advanced terms, see the complete Linksys glossary at http://www.linksys.com/glossary. Access Point - A device that allows wireless-equipped computers and other devices to communicate with a wired network. Also used to expand the range of a wireless network. Ad-hoc - A group of wireless devices communicating directly with each other (peer-to-peer) without the use of an access point. AES (Advanced Encryption Standard) - A security method that uses symmetric 128-bit block data encryption. Bandwidth - The transmission capacity of a given device or network. Bit - A binary digit. Boot - To start a device and cause it to start executing instructions. Broadband - An always-on, fast Internet connection. Browser - An application program that provides a way to look at and interact with all the information on the World Wide Web. Byte - A unit of data that is usually eight bits long Cable Modem - A device that connects a computer to the cable television network, which in turn connects to the Internet. Daisy Chain - A method used to connect devices in a series, one after the other. DDNS (Dynamic Domain Name System) - Allows the hosting of a website, FTP server, or e-mail server with a fixed domain name (e.g., www.xyz.com) and a dynamic IP address. Default Gateway - A device that forwards Internet traffic from your local area network. DHCP (Dynamic Host Configuration Protocol) - A networking protocol that allows administrators to assign temporary IP addresses to network computers by "leasing" an IP address to a user for a limited amount of time, instead of assigning permanent IP addresses.
DMZ (Demilitarized Zone) - Removes the Router's firewall protection from one PC, allowing it to be "seen" from the Internet. DNS (Domain Name Server) - The IP address of your ISP's server, which translates the names of websites into IP addresses. Domain - A specific name for a network of computers. Download - To receive a file transmitted over a network. DSL (Digital Subscriber Line) - An always-on broadband connection over traditional phone lines. Dynamic IP Address - A temporary IP address assigned by a DHCP server. EAP (Extensible Authentication Protocol) - A general authentication protocol used to control network access. Many specific authentication methods work within this framework. Encryption - Encoding data transmitted in a network. Ethernet - IEEE standard network protocol that specifies how data is placed on and retrieved from a common transmission medium. Firewall - A set of related programs located at a network gateway server that protects the resources of a network from users from other networks. Firmware - The programming code that runs a networking device. FTP (File Transfer Protocol) - A protocol used to transfer files over a TCP/IP network. Full Duplex - The ability of a networking device to receive and transmit data simultaneously. Gateway - A device that interconnects networks with different, incompatible communications protocols. Half Duplex - Data transmission that can occur in two directions over a single line, but only one direction at a time. HTTP (HyperText Transport Protocol) - The communications protocol used to connect to servers on the World Wide Web. Infrastructure - A wireless network that is bridged to a wired network via an access point. IP (Internet Protocol) - A protocol used to send data over a network.
TKIP (Temporal Key Integrity Protocol) - a wireless encryption protocol that provides dynamic encryption keys for each packet transmitted. Topology - The physical layout of a network. TX Rate - Transmission Rate. Upgrade - To replace existing software or firmware with a newer version. Upload - To transmit a file over a network. URL (Uniform Resource Locator) - The address of a file located on the Internet. VPN (Virtual Private Network) - A security measure to protect data as it leaves one network and goes to another over the Internet. WAN (Wide Area Network)- The Internet. WEP (Wired Equivalent Privacy) - A method of encrypting network data transmitted on a wireless network for greater security. WLAN (Wireless Local Area Network) - A group of computers and associated devices that communicate with each other wirelessly. WPA (Wi-Fi Protected Access) - a wireless security protocol using TKIP (Temporal Key Integrity Protocol) encryption, which can be used in conjunction with a RADIUS server.
Appendix C: Specifications
Model Standards Voice Codecs Ports Cabling Type LEDs PAP2T SIP v2 Session Initiation Protocol v2 (RFC 3261, 3262, 3263, 3264) G.729 A, G.711 a-law, G.711-law, G.726, G.723.1 Power, Internet, Phone 1, Phone 2 CAT 5 (RJ45) for Ethernet Port and RJ11 for Phone Ports Power, Internet, Phone 1, Phone 2
Maximum Ringer Load 3 REN Ring Frequency FXS Port Impedance Ring Voltage Security Features Dimensions (W x H x D) Unit Weight Power Certifications Operating Temp.
10-40 Hz Eight Configurable Settings Including North America 600 ohms, European CTR21 40-55 Vrms Configurable Password Protected Administration 3.98" x 3.98" x 1.1" (101mm x 101mm x 28mm) 5.4 oz. (0.15 kg) 5 V DC, 2.0 A FCC, CE Mark 32~113F (0~45C)
Storage Temp. Operating Humidity Storage Humidity
-77~158F (-25~70C) 10~90% Non-Condensing 5~90% Non-Condensing
Appendix D: Warranty Information
Your service provider warrants to You that, for a period of one year (the Warranty Period), your Linksys Product will be substantially free of defects in materials and workmanship under normal use. Your exclusive remedy and your service providers entire liability under this warranty will be for your service provider at its option to repair or replace the Product or refund Your purchase price less any rebates. This limited warranty extends only to the original purchaser. If the Product proves defective during the Warranty Period call your service providers Technical Support in order to obtain a Return Authorization Number, if applicable. BE SURE TO HAVE YOUR PROOF OF PURCHASE ON HAND WHEN CALLING. If You are requested to return the Product, mark the Return Authorization Number clearly on the outside of the package and include a copy of your original proof of purchase. RETURN REQUESTS CANNOT BE PROCESSED WITHOUT PROOF OF PURCHASE. You are responsible for shipping defective Products to your service provider. Your service provider pays for UPS Ground shipping from your service provider back to You only. Customers located outside of the United States of America and Canada are responsible for all shipping and handling charges. ALL IMPLIED WARRANTIES AND CONDITIONS OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE ARE LIMITED TO THE DURATION OF THE WARRANTY PERIOD. ALL OTHER EXPRESS OR IMPLIED CONDITIONS, REPRESENTATIONS AND WARRANTIES, INCLUDING ANY IMPLIED WARRANTY OF NON-INFRINGEMENT, ARE DISCLAIMED. Some jurisdictions do not allow limitations on how long an implied warranty lasts, so the above limitation may not apply to You. This warranty gives You specific legal rights, and You may also have other rights which vary by jurisdiction. This warranty does not apply if the Product (a) has been altered, except by your service provider, (b) has not been installed, operated, repaired, or maintained in accordance with instructions supplied by your service provider, or (c) has been subjected to abnormal physical or electrical stress, misuse, negligence, or accident. In addition, due to the continual development of new techniques for intruding upon and attacking networks, your service provider does not warrant that the Product will be free of vulnerability to intrusion or attack. TO THE EXTENT NOT PROHIBITED BY LAW, IN NO EVENT WILL YOUR SERVICE PROVIDER BE LIABLE FOR ANY LOST DATA, REVENUE OR PROFIT, OR FOR SPECIAL, INDIRECT, CONSEQUENTIAL, INCIDENTAL OR PUNITIVE DAMAGES, REGARDLESS OF THE THEORY OF LIABILITY (INCLUDING NEGLIGENCE), ARISING OUT OF OR RELATED TO THE USE OF OR INABILITY TO USE THE PRODUCT (INCLUDING ANY SOFTWARE), EVEN IF YOUR SERVICE PROVIDER HAS BEEN ADVISED OF THE POSSIBILITY OF SUCH DAMAGES. IN NO EVENT WILL YOUR SERVICE PROVIDERS LIABILITY EXCEED THE AMOUNT PAID BY YOU FOR THE PRODUCT. The foregoing limitations will apply even if any warranty or remedy provided under this Agreement fails of its essential purpose. Some jurisdictions do not allow the exclusion or limitation of incidental or consequential damages, so the above limitation or exclusion may not apply to You. Please direct all inquiries to your service provider.
The telephone company may make changes in its facilities, equipment, operations, or procedures that could affect the operation of the equipment. If this happens, the telephone company will provide advance notice in order for you to make the necessary modifications in order to maintain uninterrupted service. In the event this equipment should fail to operate properly, disconnect the unit from the telephone line. Try using another FCC approved device in the same telephone jack. If the trouble persists, call the telephone company repair service bureau. If the trouble does not persist and appears to be with this unit, disconnect the unit from the telephone line and discontinue use of the unit until it is repaired. Please note that the telephone company may ask that you disconnect the equipment from the telephone network until the problem has been corrected or until you are sure that the equipment is not malfunctioning. The user must use the accessories and cables supplied by the manufacturer to get optimum performance from the product. No repairs may be done by the customer. If trouble is experienced with this equipment, please contact your authorized support provider for repair and warranty information. If the trouble is causing harm to the telephone network, the telephone company may request you remove the equipment from the network until the problem is resolved. This equipment cannot be used on telephone company provided coin service. Connection to Party Line Service is subject to state tariffs. Safety Notices Caution: To reduce the risk of fire, use only No.26 AWG or larger telecommunication line cord. Do not use this product near water, for example, in a wet basement or near a swimming pool. Avoid using this product during an electrical storm. There may be a remote risk of electric shock from lightning. Industry Canada (Canada) This device complies with Industry Canada ICES-003 rule. Cet appareil est conforme la norme NMB003 d'Industrie Canada. IC Statement Operation is subject to the following two conditions: 1. This device may not cause interference and 2. This device must accept any interference, including interference that may cause undesired operation of the device.
Rglement dIndustry Canada Le fonctionnement est soumis aux conditions suivantes : 1. Ce priphrique ne doit pas causer d'interfrences; 2. Ce priphrique doit accepter toutes les interfrences reues, y compris celles qui risquent d'entraner un fonctionnement indsirable. User Information for Consumer Products Covered by EU Directive 2002/96/EC on Waste Electric and Electronic Equipment (WEEE) This document contains important information for users with regards to the proper disposal and recycling of Linksys products. Consumers are required to comply with this notice for all electronic products bearing the following symbol:
For more information, visit www.linksys.com.
Appendix F: Contact Information
Internet Phone Service Provider
Contact your service provider if the warranty support issues are to be handled by them as per your service agreements.
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Appendix F: Contact Information Internet Phone Service Provider
The PHONE ADAPTER is configurable in many ways such that it can provide a wide range of customizable services and operate in many diverse environments with a variety different vendors SIP Proxy Servers, VoIP Gateways, Voice Mail Servers, NAT applications, etc. Provisioning is the process by which the PHONE ADAPTER obtains a set of configuration parameters in order for it to operate in the Service Providers network. The complete set of configuration parameters for an PHONE ADAPTER corresponding to an individual subscriber is referred to as a configuration profile or simply a Profile. The Profile can be encoded as an XML file or a simple plain text file with a list of tag/value pairs. When the PHONE 2004 Linksys Proprietary (See Copyright Notice on Page 2) 9
ADAPTER unit is shipped from the factory, it contains a default common Profile and is considered Unprovisioned. To save costs and expedite delivery, however, it is very desirable that an Unprovisioned unit can be shipped directly from the factory to the subscribers location without any preprocessing by the Service Provider. The PHONE ADAPTER contacts the Service Providers provisioning server via the IP network or Internet when it is plugged into the subscribers home or business Local Area Network (LAN) assuming the provisioning server is reachable from the subscribers home network to pull the designated profile to be installed in that particular PHONE ADAPTER unit. Furthermore, the PHONE ADAPTER unit will periodically contact the provisioning server to download an updated profile. The protocol for downloading the configuration profile can be clear text TFTP or HTTP data or it can be encrypted TFTP, HTTP or HTTPS data if security is required. Security will be discussed in more details in a later section. This type of autonomous remote provisioning, where the individual PHONE ADAPTER unit pulls the profile from the provisioning server is very scalable and flexible. Using this provisioning method, a large number of PHONE ADAPTER units can be provisioned simultaneously and updated periodically. However, some basic information must be provided to the PHONE ADAPTER before it can be provisioned in this fashion: a) the IP address or domain name of the provisioning server to contact, and b) an ID and/or a password to send to the provisioning server such that it can associate it with a specific subscriber and obtain the corresponding profile. This information can be sent out-of-band to the subscriber via secured email or in a letter inside a welcome kit, for example. The subscriber might need to punch in some numbers using a telephone connected to the PHONE ADAPTER in order to enter this information into the unit. The PHONE ADAPTER provides an easy-to-use interface with audio instructions to make this initial configuration process as painless as possible. An alternative is for the unit to be provisioned with this basic information by the Service Provider before the unit is shipped to the subscriber. In addition to the batch mode of remote provisioning, the PHONE ADAPTER allows an interactive mode of local provisioning. One way to offer this feature is through the use of an IVR system (accessed through an attached telephone set). The user can access a diagnostic or configuration menu to check the status of the device or to change some of the settings. This method of provisioning may be applied by an administrator when the device is at the Service Providers office, or by the subscriber under the guidance of trained personnel during over-the-phone troubleshooting. A third method of entering provisioning information into the PHONE ADAPTER is by way of its integral web server via a browser on a PC. The subscriber has the option to set and adjust configuration parameters via an easy-to-use, password protected graphical user interface. This method of provisioning might be preferred by administrators who wish to access the PHONE ADAPTER over a secure corporate/institutional LAN or by the residential subscriber who is a power user.
telephone number to forward calls to. The PHONE ADAPTER provides audio instructions to prompt the user for a forwarding number and confirms that the requested service has been activated. Call FWD Unconditional All calls are immediately forwarded to the designated forwarding number. The PHONE ADAPTER will not ring or provide call waiting when Call FWD Unconditional is activated. Call FWD Busy Calls are forwarded to the designated forwarding number if the subscribers line is busy because of the following; Primary line already in a call, primary and secondary line in a call or conference. Call FWD - No Answer Calls are forwarded to the designated forwarding number after a configurable time period elapses while the PHONE ADAPTER is ringing and does not answer. 184.108.40.206.12. Anonymous Call Blocking By setting the corresponding configuration parameter on the PHONE ADAPTER, the subscriber has the option to block incoming calls that do not reveal the callers Caller ID. 220.127.116.11.13. Distinctive / Priority Ringing The PHONE ADAPTER supports a number of ringing and call waiting tone patterns to be played when incoming calls arrive. The choice of alerting pattern to use is carried in the incoming SIP INVITE message inserted by the SIP Proxy Server (or other intermediate application server in the Service Providers domain). 18.104.22.168.14. Speed Dialing The PHONE ADAPTER supports speed dialing of up to eight (8) phone numbers or IP addresses. To enter a telephone number speed dial using a touch tone telephone, the user dials a feature code (*74), followed by a number (2-9), then the destination speed dialed target number. When the user wishes to speed dial a target number, they press the corresponding speed dial assigned number followed by the # (pound) key. Users may also enter/review speed dials from User1/User2 web-pages. This interface or similar is required to enter IP address targets. 22.214.171.124. PSTN Interworking
The PHONE ADAPTER is designed to provide a transparent interworking relationship with the PSTN. Service providers can deploy the PHONE ADAPTER in such a way that PSTN endpoints wired or wireless communicating with PHONE ADAPTER endpoints do so without modification to their configuration or network settings. The service provider may choose to deploy a multi-protocol VoIP network, much the same way the PSTN supports multiple signaling schemes today. Most telecommunication providers operate equipment that supports CAS or channel associated signaling, ISDN signaling and SS7 signaling. When VoIP is introduced or used in the telecommunications landscape, it is likely that the service provider will implement a signaling gateway that supports multiple IP Telephony protocols along with legacy PSTN protocols. The signaling gateway is commonly referred to as a Softswitch. Architecture and functionality can vary greatly amongst the different softswitch vendors. The protocols used will depend on the types of connections that will be set-up across the service providers network. If the provider is simply providing transport of calls to/from their network to another providers network, but not originating or terminating calls with the endpoints, SIP will likely be used for softswitch to softswitch communication.
If the service provider is offering origination and/or termination on endpoint equipment then it is very likely that the softswitch chosen for network operations will support multiple PSTN and VoIP signaling protocols. The table below lists the most commonly accepted, de-facto standards used when implementing a VoIP signaling scheme based on the type of gateway or endpoint equipment being deployed: VoIP Equipment Type Trunking Gateways Access Gateways PBX/KTS Platforms PBX/KTS Telephone Sets Phone Adapters and IP Centrex Phones Typical Port Density Greater Than 500 Ports Between five and 500 Ports Between ten and 500 Ports One Port Up to four Ports De-Facto Signaling Standards H.248-Megaco / MGCP / IPDC SIP / H.323 SIP / H.323 / SCCP SIP / MGCP / SCCP SIP / MGCP
The PHONE ADAPTER supports SIP today. It has the capability to communicate with a variety of endpoints and signaling entities via SIP messages.
Network Address Translation (NAT) Traversal
What is a NAT or NAPT (Network Address Port Translator)?
A NAT allows multiple devices to share the same external IP address to access the resources on the external network. The NAT device is usually available as one of the functions performed by a router that routes packets between an external network and an internal (or private) one. A typical application of a NAT is to allow all the devices in a subscribers home network to access the Internet through a router with a single public IP address assigned by the ISP. The IP header of the packets sent from the private network to the public network can be substituted by the NAT with the public IP address and a port selected by the router according to some algorithm. In other words, recipient of the packets on the public network will perceive the packets as coming from the external address instead of the private address of the device where the packets are originated. In most Internet protocols, the source address of a packet is also used by the recipient as the destination to send back a response. If the source address of the packets sent from the private network to the public network is not modified by the router, the recipient may not be able to send back a response to the originator of the message since its private source IP address/port is not usable. When a packet is sent from a device on the private network to some address on the external network, the NAT selects a port at the external interface from which to send the packet to the destination address/port. The private address/port of the device, the external address/port selected by the NAT to send the packet, and the external destination address/port of the packet form a NAT Mapping. The mapping is created when the device first sends a packet from the particular source address/port to the particular destination address/port and is remembered by the NAT for a short period of time. This period varies widely from vendor to vendor; it could be a few seconds, or a few minutes, or more, or less. While the mapping is in effect, packets sent from the same private source address/port to the same public destination address/port is reused by the NAT. The expiration time of a mapping is extended whenever a packet is sent from the corresponding source to the corresponding destination. More importantly, packets sent from that public address/port to the external address/port of the NAT will be routed back to the private address/port of the mapping session that is in effect. Some NAT devices actually reuse the same mapping for the same private source address/port to any external IP address/port and/or will route packets sent to its external address/port of a mapping from any external 2004 Linksys Proprietary (See Copyright Notice on Page 2) 15
Using the Supplemental Profile Compiler
Once a plain-text file has been generated with the desired parameter settings, it needs to be compiled into a binary CFG file. The profile compiler can generate a generic unencrypted CFG file, a targeted CFG file (encrypted for a unique PHONE ADAPTER), a generic key encrypted CFG file, or a targeted and key encrypted CFG file. A generic CFG file (non-targeted) is accepted as valid by any PHONE ADAPTER device. A targeted CFG file is only accepted as valid by the PHONE ADAPTER device bearing the target MAC address. Targeted CFG files are encrypted with a 128-bit algorithmically generated key, and therefore do not require a key to be issued explicitly. Targeted CFG files provide a basic level of security for remotely locking an otherwise unprovisioned PHONE ADAPTER. The binary configuration format supports RC4 and AES symmetric key algorithms, with keys of up to 256 bits. The key can be specified explicitly as a hex-string, or it can be generated from a password or a quoted pass-phrase. In the case of passwords and pass-phrases, the internally generated key is 128 bits in length. The following command-line syntax generates a generic and unencrypted CFG file: spc pap2.txt pap2.cfg A targeted CFG file (with basic encryption) is specified by supplying the MAC address of the target device: spc -target 000e08aaa010 pap2.txt pap2.cfg
An encrypted CFG file requires either a password (or quoted pass-phrase) or a hex-string. The following lines illustrate command-line invocations for various combinations of keys and algorithms. spc spc spc spc -rc4 -aes -aes -aes -ascii-key apple4sale pap2.txt pap2.cfg -ascii-key lucky777 pap2.txt pap2.cfg -ascii-key my secret phrase pap2.txt pap2.cfg -hex-key 8d23fe7.a5c29 pap2.txt pap2.cfg
A CFG file can be both targeted and key encrypted, as suggested by the following example: spc -target 000e08aaa010 -aes -hex-key 9a20.eb47 a.txt a.cfg The status messages printed by spc can be suppressed with the --quiet command line option. Or they can be redirected to a file, with the --log file_name command line option. In the latter case, the spc command line invocation itself is also printed in the log file, preceded by a timestamp. spc -quiet. spc -log prov.log.
Encrypting and Compressing XML configuration files
Profile Example Scenarios: Enterprise LAN with DHCP Supplied TFTP Server Name: The DHCP server automatically advertises a TFTP server name to service the local network. Each PHONE ADAPTER in the network is supplied a unique CFG file based on its MAC address. The TFTP server would also contain a generic Phone Adapter2000.cfg in its tftp-root directory that contains the Profile_Rule indicated below. It would additionally carry individualized CFG files, one per device, within a tree below the tftp-root node. Each of these files would then individualize the devices. /profiles/$MA/pap2.cfg When first powered-on, unprovisioned devices would download the /pap2.cfg file from the TFTP server indicated by DHCP, (following their manufacturing default setting for the Profile_Rule parameter). The downloaded file would then direct the PHONE ADAPTER to resync to the server and fetch the individualized CFG file, as per the rule above, which completes the provisioning sequence.
VoIP Service Provider: Conceptually, a service provider solution would follow the steps as in the above example. In addition, it would then proceed to enable stronger encryption by implementing one more provisioning step, with one more level of redirection, involving a random CFG file path and encryption key. Hence, each of the first-stage CFG files above would point to a second-stage CFG file, with entries such as the following:
Profile_Rule [--key $B] ps.global.com/profiles/active/$A/pap2.cfg; GPP_A Dz3P2q9sVgx7LmWbvu; GPP_B 83c1e792bc6a824c0d18f429bea52d8483f2a24b32d75bc965d05e38c163d5ef; In practice, the first provisioning stage (which individualizes each PHONE ADAPTER into fetching a unique CFG file) could be preconfigured during manufacturing. For added security, the second stage, which introduces strong encryption, may be performed inhouse, prior to shipping an PHONE ADAPTER to each end-user. Release 2.0 supports SSL-based key exchanges, alleviating the need for this in-house step, while preserving strong security for the provisioning process. A provisioning flow chart, from the point of view of the PHONE ADAPTER endpoint is presented in a later section. Log Resync Request Message: ParName: Default: Log_Resync_Request_Msg $PN $MAC - Requesting resync $SCHEME://$SERVIP:$PORT$PATH
The Log_Resync_Request_Msg is a script that defines the message sent to the configured Syslog server whenever the PHONE ADAPTER attempts to resync with the provisioning server. The string supports one level of macro substitution, with the same variables as for the Profile_Rule above. An empty string does not generate a syslog message.
600@16;1(.25/.25/1)" 985@-16,1428@16,1777@16;20(.380/0/1,.380 /0/2,.380/0/3,0/4/0) 914@-16,1371@16,1777@16;20(.274/0/1,.274 /0/2,.380/0/3,0/4/0) 914@-16,1371@16,1777@16;20(.380/0/1,.380 /0/2,.380/0/3,0/4/0) 985@-16,1371@16,1777@16;20(.380/0/1,.274 /0/2,.380/0/3,0/4/0) 350@-19,440@19;2(.1/.1/1+2);10(* /0/1+2) 350@-19,440@19;2(.2/.2/1+2);10(* /0/1+2) 600@16;*(.1/.1/1,.1/.1/1,. 1/9.5/1) 350@16;30(.1/.1/1,.1/9.7/ 1) 397@-19,507@19;15(0/2/0,.2/.1/1,. 1/2.1/2)
See <SIT1 Tone>
See <SIT 1 Tone>
MWI Dial Tone
Cfwd Dial Tone
This tone is played instead of <Dial Tone> when there are unheard messages in the subscribers mail box Special dial tone played when call forward all is activated Indicate to the local user that the far end has placed the call on hold Plays to all parties when a 3-way conference is in progress This tone is played when a call is successfully switched to secure mode. It should be played only for a short while (< 30s) and at a reduced level (< -19 dBm) so that it will not interfere with the conversation.
Secure Call Indication Tone
Notes: 1. Reorder Tone is played automatically when <Dial Tone> or any of its alternatives times out 2. Off Hook Warning Tone (also called Howler Tone) is played when Reorder Tone times out
4.10. Less Frequently Used Paramters
Advanced Protocol Parameters
Description Type Default
Parameter Name SIP Parameters Max Forward
SIP Max-Forward value. Range: 1 255
Number of times to allow an INVITE to be redirected by a 3xx response to avoid an infinite loop.
Note: This parameter currently has no effect: there is no limit on number of redirection.
Max Auth SIP User Agent Name SIP Server Name
SIP Accept Language Remove Last Reg Use Compact Header
Maximum number of times a request may be challenged (0-255) User-Agent Header to be used by the unit in outbound requests. If empty, the header is not included. Server Header to used by the unit in responses to inbound responses. If empty, the header is not included. Accept-Language Header to be used by the unit. If empty, the header is not included. Remove last registration before registering a new one if value is different one. If set to yes, the PHONE ADAPTER will use compact SIP headers in outbound SIP messages. If set to no the PHONE ADAPTER will use normal SIP headers.
2 Linksys/ $version Linksys/ $version
SIP Timer Values (sec) SIP T1 RFC 3261 T1 value (RTT Estimate). Range: sec SIP T2 RFC 3261 T2 value (Maximum retransmit interval for non-INVITE requests and INVITE responses). Range: sec SIP T4 RFC 3261 T4 value (Maximum duration a message will remain in the network). Range: sec SIP Timer B INVITE time out value. Range: sec SIP Timer F Non-INVITE time out value. Range: sec SIP Timer H INVITE final response time out value. Range: sec SIP Timer D ACK hang around time. Range: sec SIP Timer J Non-INVITE response hang around time. Range: sec INVITE Expires INVITE request Expires header value in sec. 0 = do not include Expires header in INVITE. 31 Range: 0 (2 1) ReINVITE Expires ReINVITE request Expires header value in sec. 0 = do not include Expires header in the 31 request. Range: 0 (2 1) Reg Min Expires Minimum registration expiration time allowed from the proxy in the Expires header or as a Contact header parameter. If proxy returns something less this value, then the minimum value is used. Reg Max Expires Maximum registration expiration time allowed from the proxy in the Min-Expires header. If 2004 Linksys Proprietary (See Copyright Notice on Page 2)
Time3 Time3 Time3 Time3 Time3 Time0
Reg Retry Intvl
Reg Retry Long Interval
value is larger than this, then the maximum value is used Interval to wait before the PHONE ADAPTER retries registration again after encountering a failure condition during last registration When Registration fails with a SIP response code that does no match <Retry Reg RSC>, the PHONE ADAPTER will wait for the delay specified in this parameter before retrying. If this parameter is 0, the PHONE ADAPTER will stop retrying. This value should be much larger than <Reg Retry Intvl> which should not be 0.
Response Status Code Handling 1 SIT1 RSC SIP response status code to INVITE on which to play the SIT1 Tone 1 SIT2 RSC SIP response status code to INVITE on which to play the SIT2 Tone 1 SIT3 RSC SIP response status code to INVITE on which to play the SIT3 Tone 1 SIT4 RSC SIP response status code to INVITE on which to play the SIT4 Tone Try Backup RSC SIP response status code on which to retry a backup server for the current request Retry Reg RSC Interval to wait before the PHONE ADAPTER retries registration again after encountering a failure condition during last registration RTP Parameters 2 RTP Port Min RTP Port Max
RscTmplt RscTmplt RscTmplt RscTmplt RscTmplt Time0 30
RTP Packet Size RTCP Tx Interval4
Minimum port number for RTP transmission and reception Maximum port number for RTP transmission and reception Packet size in sec. Valid values must be multiple of 0.01s. Range: 0.01 0.16 Controls the interval (sec) to send out RTCP sender report on an active connection. Range: (s)
Port Port Time3 Time0
Notes: 1. Reorder or Busy Tone will be played by default for all unsuccessful response status code 2. <RTP Port Min> and <RTP Port Max> should define a range that contains at least 4 even number ports, such as 3. If inbound SIP requests contain compact headers, PHONE ADAPTER will reuse the same compact headers when generating the response regardless the settings of the <Use Compact Header> parameter. If inbound SIP requests contain normal headers, PHONE ADAPTER will substitute those headers with compact headers (if defined by RFC 261) if <Use Compact Header> parameter is set to yes. 4. During an active connection, the PHONE ADAPTER can be programmed to send out compound RTCP packet on the connection. Each compound RTP packet except the last one contains a SR (Sender Report) and a SDES.(Source Description). The last RTCP packet contains an additional BYE packet. Each SR except the last one contains exactly 1 RR (Receiver Report); the last SR 2004 Linksys Proprietary (See Copyright Notice on Page 2)
Parameter Name Line Enable 2 MOH Server
SIP Port SIP TOS/DiffServ Value RTP TOS/DiffServ Value 3 SAS Enable
Port Byte Byte Bool
5060 0x68 0xb8 No
SAS DLG Refresh 3 Intvl
SAS Inbound RTP 3 Sink
SIP Debug Option Network Jitter Level
SIP 100REL Enable
Blind Attn-Xfer Enable
SAS lines own IP address is used in the c = line and a=sendrecv. In that case the SAS client will stream RTP packets to the SAS line. The default value is [empty]. None, 1-line, full, exclude OPTIONS, exclude REGISTER, exclude NOTIFY, 4 settings are available: very high, high, medium, low. This parameter affects how jitter buffer size is adjusted in the PHONE ADAPTER. Jitter buffer size is adjusted dynamically. The minimum jitter buffer size is 30 ms or (10 ms + current RTP frame size), which ever is larger, for all jitter level settings. But the starting jitter buffer size value is larger for higher jitter levels. This parameter controls the rate at which to adjust the jitter buffer size to reach the minimum. If the jitter level is set to high, then the rate of buffer size decrement is slower (more conservative), else faster (more aggressive). Enable the support or the 100rel SIP extension for reliable transmission of provisional responses (18x) and the use of PRACK requests. If enabled, the PHONE ADAPTER performs an attended transfer operation by terminating the current call leg, and blind transferring the other call leg. If disabled, the PHONE ADAPTER performs an attended transfer by referring the other call leg to the current call leg while maintaining both call legs.
Notes: 1. If proxy responded to REGISTER with a smaller Expires value, the PHONE ADAPTER will renew registration based on this smaller value instead of the configured value. If registration failed with an Expires too brief error response, the PHONE ADAPTER will retry with the value given in the MinExpires header in the error response. 2. MOH Notes: The remote party must indicate that it can receive audio while holding MOH to work. That is the SIP 2xx response from the remote party in reply to the re-INVITE from the PHONE ADAPTER to put the call on hold must have the SDP indicate a sendrecv or recvonly attribute and the remote destination address and port must not be 0 3. SAS Notes: Either or both of lines 1 and 2 can be configured as an SAS server. Each server can maintain up to 5 simultaneous calls. If the second line on the PHONE ADAPTER is disabled, then the SAS line can maintain up to 10 simultaneous calls. Further incoming calls will receive a busy signal (SIP 486 Response). The streaming audio source must be off-hook for the streaming to occur. Otherwise incoming calls will get a error response (SIP 503 Response). The SAS line will not ring for incoming calls even if the attached equipment is on-hook If no calls are in session, battery is removed from tip-and-ring of the FXS port. Some audio source devices have an LED to indicate the battery status. This can be used as a visual indication whether any audio streaming is in progress.
instructions to listen to your messages. Expected Call and Network Behavior When voice mail is available for a subscriber, a notification message will be sent from the Voice Mail server to the PHONE ADAPTER. When the user dials their own phone number, the PHONE ADAPTER connects the subscriber their voice mail system which can then connect them to their individual voice mail box. Follow instructions of the voice mail system or simply hang-up the telephone.
5.10. Attendant Call Transfer
Service Description Attendant Call Transfer lets a customer use their Touchtone phone to send a call to any other phone, inside or outside their business, including a wireless phones. While in a call with the party to be transferred: Press the switch hook or flash button on the phone to place the party on hold Listen for three short tones followed by dial tone Dial the number to which you will transfer the caller Stay on the line until the called number answers Announce the call Press the switch hook or flash button adding the held party to the call Hang up to connect the two parties and transfer the call Note: You can hook flash while the 3 party is ringing to start an early conference. Then hang up to complete the transfer without waiting for rd the 3 party to answer first. Expected Call and Network Behavior When the user presses the switch hook or flash button, the transferee is placed on hold. When the user successfully dials the transfer number and the party answers the transferee can be added to the call by pressing the switch hook or flash button creating a three-way conference. When the user hangs up the phone the transferee and the called party
remain in a call. User Action Required to Deactivate or End Not applicable.
5.11. Unattended or Blind Call Transfer
Service Description Unattended or Blind Call Transfer lets a customer use their Touchtone phone to send a call to any other phone, inside or outside their business, including a wireless phones. While in a call with the party to be transferred: Press the switch hook or flash button on the phone to place the party on hold Enter *__ Dial the number to which you will transfer the caller The call is transferred when a complete number is entered. You will hear a short confirmation tone, followed by regular dial tone Expected Call and Network Behavior When the user presses the switch hook or flash button, the transferee is placed on hold. When the user successfully dials the transfer number, the transferee will automatically call the dialed number. No applicable.
Codec Name Assignment
Negotiation of the optimal voice codec is sometimes dependent on the PHONE ADAPTER devices ability to match a codec name with the far-end device/gateway codec name. The PHONE ADAPTER allows the network administrator to individually name the various codecs that are supported such that the correct codec successfully negotiates with the far end the equipment.
A user (if enabled by service provider or administrator) has the option to make an outbound call secure in the sense that the audio packets in both directions are encrypted.
G.711 (A-law and m-law)
This very low complexity codec supports uncompressed 64 kbps digitized voice transmission at one through ten 5 ms voice frames per packet. This codec provides the highest voice quality and uses the most bandwidth of any of the available codecs. 126.96.36.199. G.726
This low complexity codec supports compressed 16, 24, 32 and 40 kbps digitized voice transmission at one through ten 10 ms voice frames per packet. This codec provides the high voice quality. 188.8.131.52. G.729A
The ITU G.729 voice coding algorithm is used to compress digitized speech. Linksys supports G.729. G.729A is a reduced complexity version of G.729. It requires about half the processing power to code G.729. The G.729 and G.729A bit streams are compatible and interoperable, but not identical. 184.108.40.206. G.723.1
The PHONE ADAPTER supports the use of ITU G.723.1 audio codec at 6.4 kbps. Up to 2 channels of G.723.1 can be used simultaneously. For example, Line 1 and Line 2 can be using G.723.1 simultaneously, or Line 1 or Line 2 can initiate a 3-way conference with both call legs using G.723.1.
The administrator can select which low-bit-rate codec to be used for each line. G711a and G711u are always enabled.
When no static payload value is assigned per RFC 1890, the PHONE ADAPTER can support dynamic payloads for G.726.
Adjustable Audio Frames Per Packet
This feature allows the user to set the number of audio frames contained in one RTP packet. Packets can be adjusted to contain from audio frames. Increasing the number of packets decreases the bandwidth utilized but it also increases delay and may affect voice quality.
Dial_Plan "(*xx|11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)" ; Enable_IP_Dialing "No" ; # *** FXS Port Polarity Configuration Idle_Polarity Caller_Conn_Polarity Callee_Conn_Polarity # *** Call Forward Settings Cfwd_All_Dest Cfwd_Busy_Dest Cfwd_No_Ans_Dest Cfwd_No_Ans_Delay ! ! ! ! "" ; "" ; "" ; "20" ; "Forward" ; "Forward" ; "Forward" ; # options: Forward/Reverse # options: Forward/Reverse # options: Forward/Reverse
# *** Selective Call Forward Settings Cfwd_Sel1_Caller Cfwd_Sel1_Dest Cfwd_Sel2_Caller Cfwd_Sel2_Dest Cfwd_Sel3_Caller Cfwd_Sel3_Dest Cfwd_Sel4_Caller Cfwd_Sel4_Dest Cfwd_Sel5_Caller Cfwd_Sel5_Dest Cfwd_Sel6_Caller Cfwd_Sel6_Dest Cfwd_Sel7_Caller Cfwd_Sel7_Dest Cfwd_Sel8_Caller Cfwd_Sel8_Dest Cfwd_Last_Caller Cfwd_Last_Dest Block_Last_Caller Accept_Last_Caller # *** Speed Dial Settings Speed_Dial_2 Speed_Dial_3 Speed_Dial_4 Speed_Dial_5 Speed_Dial_6 Speed_Dial_7 Speed_Dial_8 Speed_Dial_9 ! ! ! ! ! ! ! ! "" "" "" "" "" "" "" "" ; ; ; ; ; ; ; ; ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! "" "" "" "" "" "" "" "" "" "" "" "" "" "" "" "" "" "" "" "" ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ;
# *** Supplementary Service Settings CW_Setting Block_CID_Setting Block_ANC_Setting DND_Setting CID_Setting CWCID_Setting Dist_Ring_Setting Secure_Call_Setting ! ! ! ! ! ! ! "Yes" ; "No" ; "No" ; "No" ; "Yes" ; "Yes" ; "Yes" ; "No" ;
# *** Distinctive Ring Settings Ring1_Caller Ring2_Caller Ring3_Caller Ring4_Caller Ring5_Caller Ring6_Caller Ring7_Caller Ring8_Caller # *** Ring Settings Default_Ring ! "1" ; # options: 1/2/3/4/5/6/7/8 Default_CWT ! "1" ; # options: 1/2/3/4/5/6/7/8 Hold_Reminder_Ring ! "8" ; # options: 1/2/3/4/5/6/7/8/none Call_Back_Ring ! "7" ; # options: 1/2/3/4/5/6/7/8 Cfwd_Ring_Splash_Len ! "0" ; Cblk_Ring_Splash_Len ! "0" ; VMWI_Ring_Splash_Len ! ".5" ; VMWI_Ring_Policy "New VM Available" ; # options: New VM Available/New VM Becomes Available/New VM Arrives Ring_On_No_New_VM "No" ; # *** Call Progress Tones "350@-19,440@-19;10(*/0/1+2)" ; Dial_Tone Second_Dial_Tone "420@-19,520@-19;10(*/0/1+2)" ; Outside_Dial_Tone "420@-16;10(*/0/1)" ; Prompt_Tone "520@-19,620@-19;10(*/0/1+2)" ; Busy_Tone "480@-19,620@-19;10(.5/.5/1+2)" ; Reorder_Tone "480@-19,620@-19;10(.25/.25/1+2)" ; Off_Hook_Warning_Tone "480@-10,620@0;10(.125/.125/1+2)" ; Ring_Back_Tone "440@-19,480@-19;*(2/4/1+2)" ; Confirm_Tone "600@-16;1(.25/.25/1)" ; SIT1_Tone "985@-16,1428@-16,1777@-16; 20(.380/0/1,.380/0/2,.380/0/3,0/4/0)" ; SIT2_Tone "914@-16,1371@-16,1777@-16; 20(.274/0/1,.274/0/2,.380/0/3,0/4/0)" ; SIT3_Tone "914@-16,1371@-16,1777@-16; 20(.380/0/1,.380/0/2,.380/0/3,0/4/0)" ; SIT4_Tone "985@-16,1371@-16,1777@-16; 20(.380/0/1,.274/0/2,.380/0/3,0/4/0)" ; MWI_Dial_Tone "350@-19,440@-19;2(.1/.1/1+2);10(*/0/1+2)" ; Cfwd_Dial_Tone "350@-19,440@-19;2(.2/.2/1+2);10(*/0/1+2)" ; Holding_Tone "600@-19;*(.1/.1/1,.1/.1/1,.1/9.5/1)" ; Conference_Tone "350@-19;20(.1/.1/1,.1/9.7/1)" ; Secure_Call_Indication_Tone "397@-19,507@-19;15(0/2/0,.2/.1/1,.1/2.1/2)" ; # *** Distinctive Ring Patterns Ring1_Cadence Ring2_Cadence Ring3_Cadence Ring4_Cadence Ring5_Cadence Ring6_Cadence Ring7_Cadence Ring8_Cadence "60(2/4)" ; "60(.3/.2,1/.2,.3/4)" ; "60(.8/.4,.8/4)" ; "60(.4/.2,.3/.2,.8/4)" ; "60(.2/.2,.2/.2,.2/.2,1/4)" ; "60(.2/.4,.2/.4,.2/4)" ; "60(.4/.2,.4/.2,.4/4)" ; "60(0.25/9.75)" ; ! ! ! ! ! ! ! ! "" "" "" "" "" "" "" "" ; ; ; ; ; ; ; ;
Dist_Ring_Act_Code Dist_Ring_Deact_Code Speed_Dial_Act_Code Secure_All_Call_Act_Code Secure_No_Call_Act_Code Secure_One_Call_Act_Code Secure_One_Call_Deact_Code Referral_Services_Codes Feature_Dial_Services_Codes
"*26" "*46" "*74" "*16" "*17" "*18" "*19" "" ; "" ;
; ; ; ; ; ; ;
# *** Outbound Call Codec Selection Codes Prefer_G711u_Code Force_G711u_Code Prefer_G711a_Code Force_G711a_Code Prefer_G723_Code Force_G723_Code Prefer_G726r16_Code Force_G726r16_Code Prefer_G726r24_Code Force_G726r24_Code Prefer_G726r32_Code Force_G726r32_Code Prefer_G726r40_Code Force_G726r40_Code Prefer_G729a_Code Force_G729a_Code # *** Miscellaneous Set_Local_Date_(mm/dd) "" ; Set_Local_Time_(HH/mm) "" ; Time_Zone "GMT-07:00" ; # options: GMT-12:00/ GMT-11:00/GMT-10:00/GMT-09:00/GMT-08:00/GMT-07:00/GMT-06:00/GMT-05:00/ GMT-04:00/GMT-03:30/GMT-03:00/GMT-02:00/GMT-01:00/GMT/GMT+01:00/ GMT+02:00/GMT+03:00/GMT+03:30/GMT+04:00/GMT+05:00/GMT+05:30/GMT+05:45/ GMT+06:00/GMT+06:30/GMT+07:00/GMT+08:00/GMT+09:00/GMT+09:30/GMT+10:00/ GMT+11:00/GMT+12:00/GMT+13:00 FXS_Port_Impedance "600" ; # options: 600/900/600+2.16uF/ 900+2.16uF/270+750||150nF/220+820||120nF/220+820||115nF/370+620||310nF FXS_Port_Input_Gain "-3" ; FXS_Port_Output_Gain "-3" ; DTMF_Playback_Level "-16" ; DTMF_Playback_Length ".1" ; Detect_ABCD "Yes" ; Playback_ABCD "Yes" ; Caller_ID_Method "Bellcore(N.Amer,China)" ; # options: Bellcore(N.Amer,China)/DTMF(Finland,Sweden)/DTMF(Denmark)/ETSI DTMF/ ETSI DTMF With PR/ETSI DTMF After Ring/ETSI FSK/ETSI FSK With PR(UK) FXS_Port_Power_Limit "3" ; # options: 1/2/3/4/5/6/7/8 Protect_IVR_FactoryReset "No" ; "*017110" ; "*027110" ; "*017111" ; "*027111" ; "*01723" ; "*02723" ; "*0172616" ; "*0272616" ; "*0172624" ; "*0272624" ; "*0172632" ; "*0272632" ; "*0172640" ; "*0272640" ; "*01729" ; "*02729" ;
A/D ANC B2BUA Bool CA CAS CDR CID Analog To Digital Converter Anonymous Call Back to Back User Agent Boolean Values. Specified as yes and no, or 1 and 0 in the profile Certificate Authority CPE Alert Signal Call Detail Record Caller ID
Circuits: The communication path(s) that carry calls between two points on a network. Customer Premise Equipment: The only part of the telecommunications system that the customer comes into direct contact with. Example of such pieces of equipment are: telephones, key systems, PBXs, voicemail systems and call accounting systems as well as wiring telephone jacks. The standard for this equipment is set by the FCC, and the equipment is supplied by an interconnect company. Dedicated Access: Customers have direct access to the long-distance provider via a special circuit (T1 or private lines). The circuit is hardwired from the customer site to the POP and does not pass through the LEC switch. The dial tone is provided from the long-distance carrier. Dedicated Access Line (DAL): Provided by the local exchange carrier. An access line from the customers telephone equipment directly to the long-distance companys switch or POP. Demarcation Point: This is where the LECs ownership and responsibility (wiring, equipment) ends and the customers responsibilities begin. Direct Inward Dialing (DID): Allows an incoming call to bypass the attendant and ring directly to an extension. Available on most PBX systems and a feature of Centrex service. Dual Tone Multifrequency (DTMF): Better known as the push button keypad. DTMF replaces dial pulses with electronically produced tones for network signaling. Enhanced Service: Services that are provided in addition to basic long distance and accessed by way of a touchtone phone through a series of menus. Exchange Code (NXX): The first three digits of a phone number. Flat-rate Pricing: The customer is charged one rate (sometimes two rates, one for peak and one for off-peak) rather than a mileage-sensitive program rate. IXC (Interexchange Carrier): A long-distance provider that maintains its own switching equipment. IVR (Interactive Voice Response): Provides mechanism for information to be stored and retrieved using voice and a touchtone telephone. Local Loop: The local telephone company provides the transmission facility from the customer to the telephone companys office, which is engineered to carry voice and/or data. North American Numbering Plan (NANP): How we identify telephone numbers in North America. We can identify the telephone number based on their three separate components (NPA) (NXX) (XXXX). PIN (Personal Identification Code): A customer calling/billing code for prepaid and pay-as-you-go calling cards. Private Branch Exchange: Advanced phone system commonly used by the medium to larger customer. It allows the customer to perform a variety of in-house routing (inside calling). The dial tone that is heard when the customer picks up the phone is an internal dial tone. SS7 (System Signaling Number 7): Technology used by large carriers to increase the reliability and speed of transmission between switches. Switch (Switching): Equipment that connects and routes calls and provides other interim functions such as least cost routing, IVR, and voicemail. It performs the traffic cop function of telecommunications via automated management decisions. Touchtone (DTMF): The tone recognized by a push button (touchtone) telephone. Unified Messaging: Platform that lets users send, receive, and manage all email, voice, and fax messages from any telephone, PC, or information device. Voice Mail: A system that allows storage and retrieval of voice messages through voicemail boxes. 2004 Linksys Proprietary (See Copyright Notice on Page 2)
The Linksys Phone Adapter enables high-quality feature-rich telephone service through your cable or DSL Internet connection. Just plug it into your home router or gateway and use the two standard telephone jacks to connect your existing phones or fax machines. Each phone jack operates independently, with separate phone service and phone numbers - like having two phone lines. With an appropriate Internet telephone service provider, you'll get clear telephone reception and reliable fax connections, even while using the Internet at the same time for normal data operations. With Internet telephony, along with low domestic and international phone rates, an impressive array of special phone features are available. Choose your preferred free local dialing US area code, regardless of where you live. Or add a virtual phone number in any area code, forwarded to your Internet phone. You can even add a toll-free number. The Linksys Phone Adapter is compatible with these and all of the other special telephone features that are available from your telephone service provider, such as Caller ID, Call Waiting, Voicemail, Call Forwarding, Distinctive Ring, etc. Let the Linksys Phone Adapter turn your existing Internet connection into a high-quality high-value telephone service.
|Device Type||VoIP phone adapter|
|Data Link Protocol||Ethernet, Fast Ethernet|
|Status Indicators||Port status, power|
|Compliant Standards||IEEE 802.3, IEEE 802.3u|
|VoIP Protocols||SIP v2|
|Voice Codecs||G.723.1, G.729a, G.711u, G.711a, G.726|
|Telephony Interfaces||2 phone (FXS)|
|Expansion / Connectivity|
|Interfaces||1 x network - Ethernet 10Base-T/100Base-TX - RJ-45 2 x phone line - FXS - RJ-11|
|Cables Included||1 x network cable|
|Compliant Standards||FCC Class B certified, CE, cUL|
|Power Device||Power adapter - external|
|Min Operating Temperature||41 °F|
|Max Operating Temperature||113 °F|
|Humidity Range Operating||10 - 90%|
|Universal Product Identifiers|
KV-25X5K MX3000 Dslink 260E Z713C Samsung J770 CK-300 M55-S329 PM-30 Ericsson W800 SV-DVD640 ASD-51N 999-VC EMT-10 Touch Full EYE-Q Mini 8 0 Password SRE 3000 AZ1202 RM2200 KH 6778 120 IS XR-1750 Firmware HPM-198 Photosmart M425 6339dwde Nice P200 T730SH 2-C210 Dimension C521 Logicom G303 Aviator TX-W32d2DP VS 250 Kodak C300 Notice Edition Aficio 2027 ICN510 5008MG 2920N Z8612 WX-C3010 PCR-50 Pearl 2008 Scanner Supermoto CDX-C5000RV DVR-640H-S Kombat-armageddon Suunto SK-7 ZVT860X Instruments Torq KDC-W4034AY NV-GS120EG Casio 3796 MPH100 P7500 Megane GT Samsung ES55 SD-RH85 FZS600-2000 PSR-J21 Wacs7000 Asus P4BE Aspire-5040 Travelmate-5720 MHC-RG330 LN32R71B Nuvi 5000 Icrb35 SRU5010-86 E5700 MC-7844AS AWF1210 WD-14126RD BDP-S350 LG WD SC-4795 Sbcvl1200 943NW F250-2004 SGH-X510 W3000H-BNS CQ-C1321NE 318IS ZR500 Maximus Giro Plus DMC-FZ38 FG-1025R F40HP-2005 RXD-700 21PT1542 58H Reaver 2 Speedstream 4200 Soundcraft EPM Unlock Asko 7704 XVS1300A-2008 2 1 XRS9730
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