SIP Analog Telephone Adapter VoIP Analog Telephone Adapter VIP-156 / VIP156PE / VIP-157 / VIP-157S 1
SIP Analog Telephone Adapter Copyright Copyright© 2013 by PLANET Technology Corp. All rights reserved. No part of this publication may be reproduced, transmitted, transcribed, stored in a retrieval system, or translated into any language or computer language, in any form or by any means, electronic, mechanical, magnetic, optical, chemical, manual or otherwise, without the prior written permission of PLANET.
SIP Analog Telephone Adapter or peripheral devices). Any changes or modifications not expressly approved by the party responsible for compliance could void the user’s authority to operate the equipment. This device complies with Part 15 of the FCC Rules. Operation is subject to the Following two conditions: (1) This device may not cause harmful interference, and (2) this Device must accept any interference received, including interference that may cause undesired operation.
SIP Analog Telephone Adapter TABLE OF CONTENTS Chapter 1 Introduction............................................................................................................ 6 Overview....................................................................................................................................................... 6 Package Contents.........................................................................................................................................
SIP Analog Telephone Adapter Codec Setting ..................................................................................................................................... 43 SIP Advance Setting........................................................................................................................... 44 Chapter 8 Advance Setting .................................................................................................. 49 Status Log .............................................
SIP Analog Telephone Adapter Chapter 1 Introduction 1 Overview Based on years of VoIP manufacturing experiences, PLANET Technology VoIP total solutions are known for advanced implementation of standards based telephony with mass deployment capability. Cost-effective, High-performance Cost-effective, easy-to-install and simple-to-use, the 802.3af PoE integration(VIP-156PE) converts standard telephones to IP-based networks.
SIP Analog Telephone Adapter Enhanced, Full-Featured VoIP Adapter For the new generation communication age, the ATA device supports IPV6 and VPN connection to provide users with more flexible and advantageous communication product.
SIP Analog Telephone Adapter IVR Function to Easily Identify and Manage the ATA Through the Interactive voice response (IVR) function, user can simply press some function key to search the device information or program the phone feature, e.g #120 to check the LAN IP address, #112 + xxx*xxx*xxx*xxx# to assign the LAN IP address…..
SIP Analog Telephone Adapter Features ¾ Product features Feature-rich telephone service over home or office Internet/Intranet connection Cost effectiveness, field proven compatibility, and stability Web-based and telephone keypad machine configuration Remote administrator authentication Voice prompt for machine configurations DMZ and MAC clone ¾ VoIP Feature SIP 2.0 (RFC3261) compliant Peer-to-Peer / SIP proxy calls Voice codec support: G.711, G.723.1, G.729A/G.
SIP Analog Telephone Adapter Respective models/descriptions are shown below: VIP-156: SIP Analog Telephone Adapter VIP-156PE: 802.
SIP Analog Telephone Adapter Left / Right Panel of VIP-156PE Front Panel of VIP-157 Left / Right Panel of VIP-157 11
SIP Analog Telephone Adapter Front Panel of VIP-157S Left / Right Panel of VIP-157S LED Display & Button 1 PC RJ-45 connector, to maintain the existing network structure, connected directly to the PC through straight CAT-5 cable RJ-45 connector, for Internet access, connected directly to Switch/Hub through straight CAT-5 cable. 2 LAN The LAN interface also can be connected with 802.
SIP Analog Telephone Adapter 4 Reset Reset to the factory default setting Machine default IP is http://192.168.0.1. Press RESET button on rear panel for over 5 seconds to reset the VoIP Phone Adapter to factory default value. (Except speed dial and call forward settings) LED display of VIP-156 / VIP-156PE LED Indicators Descriptions PWR Power is supplied to the device. STATUS The Status LED will flash when the machine is operational LNK/ACT OFF: the device is connected to LAN at 10Mb/s.
SIP Analog Telephone Adapter Chapter 2 Preparations & Installation 2 Physical Installation Requirements This chapter illustrates basic installation of ATA analog Phone Adapter (“ATA” in the following term) • Network cables. Use standard 10/100BaseT network (UTP) cables with RJ45 connectors. • TCP/IP protocol must be installed on all PCs.
SIP Analog Telephone Adapter Please locate your PC in the same network segment (192.168.0.x) of ATA. If you’re not familiar with TCP/IP, please refer to related chapter on user’s manual CD or consult your network administrator for proper network configurations LAN IP address configuration via web configuration interface Execute your web browser, and insert the IP address (default: 192.168.0.1) of VIP in the address bar.
SIP Analog Telephone Adapter IVR Menu Choice Machine operation Parameter(s) Notes: ATA will change to DHCP #111# #112xxx*xxx*xxx* Set DHCP client None Client Setup Static IP Address xxx# Use the * (star) key DHCP will be disabled and when entering a decimal system will change to the point. Static IP type. Use the * (star) key #113xxx*xxx*xxx* Set Network Mask xxx# when entering a decimal Must set Static IP first. point.
SIP Analog Telephone Adapter of the ATA. IVR will announce the current gateway IP #124# Check Gateway IP Address #125# Check Primary DNS Server Setting #126# Check LAN IP Address #128# Check Firmware Version address of the ATA. IVR will announce the current setting in the Primary DNS field. IVR will announce the current LAN port IP address of the ATA. IVR will announce the version of the firmware running on the ATA. The following keypad commands can be used to set up the main function .
SIP Analog Telephone Adapter Server, 2: user FTP Server #145# Forward function disable Disable forwrad funciton enable forward to FXS Eanble forward to FXS Port Port enable forward to FXO Eanble forward to FXO Port Port #116# Enable PPTP function None Enable PPTP function #117# Disable PPTP function None Disable PPTP function #118# Enable VLAN function None Enable VLAN function #119# Disable VLAN function None Disable VLAN function #146+Number# #147+Number# L Hint Please conta
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SIP Analog Telephone Adapter Chapter 3 Network Service Configurations 3 Configuring and monitoring your ATA from web browser The ATA integrates a web-based graphical user interface that can cover most configurations and machine status monitoring. Via standard web browser, you can configure and check machine status from anywhere around the world.
SIP Analog Telephone Adapter VoIP Phone Adapter main page 21
SIP Analog Telephone Adapter Chapter 4 VoIP Telephone Adapter Configurations 4 Status Status shows all the system information like WAN/LAN IP address, System information, IPV6 connection information, Register status and VPN connection message. (After you set up the VPN line, the status will start to show.
SIP Analog Telephone Adapter Phone Book ATA can set up 140 records of Phone Book. User can dial the Name records to make calls via Phone Book feature. Field Phone Book Page Description The default is Page 1. It can select Page1 ~ Page 7 to look through Phone Book records.
SIP Analog Telephone Adapter Phone Name URL The record number is from 1 ~ 140; it can set up 140 records in total. The name of Phone Book records; it only can input numerals character. Fill in the outgoing number (Line Number) or IP address. Delete Clean this item’s data. Export csv Save the phone book data as CSV file. Upload Upload the phone book file If you need to add a phone number to the Phone Book list, you need to input the position, the name, and the phone number (by URL type).
SIP Analog Telephone Adapter Call Service [Call Forward] This page defines Call Forward function. You can set up the phone number you want to forward on this page. There are three types of Forward mode. You can choose All Forward, Busy Forward, and No Answer Forward by clicking the icon. All Forward: All incoming call will forward to the number you choose. You can input the name and the phone number in the URL field.
SIP Analog Telephone Adapter All to PSTN/ No Answer to PSTN (VIP-157): the VIP-157 not only supports Call Forward to IP calls, but also can forward the calls to PSTN. You can choose the Call Forward type with PSTN, then input the name and the PSTN number in the URL/Number field. IP Line Forward function for VIP-157 The IP Line Forward function’s incoming call is IP call type, and the destination is IP or PSTN call type.
SIP Analog Telephone Adapter is larger than the “To” time, the Block time will from Day 1 to Day 2. When you finish the setting, please click the Submit button. [Alarm Type] This page defines the Alarm setting on this page. It provides the alarm function, and it can set up the Alarm Time to get the telephone ringed up every day. Alarm Type: The default is Off. If set up as On, the telephone will ring up at a specific time. Alarm Time: It can set up the system prompt time within 24 hours.
SIP Analog Telephone Adapter Volume Settings for VIP-156T/VIP-156PT Besides the above settings, VIP-157 also can set the volume of PSTN. PSTN-Out Volume is to set the PSTN volume you can hear. PSTN-In Gain is to set the volume sent out to the other side’s handset. Volume Settings for VIP-157 Dial Plan Setting This page defines the Dial Plan Setting function. This function is when you input the phone number by the keypad but you don’t need to press “#”. After time out the system will dial directly.
SIP Analog Telephone Adapter Dial Plan Settings for VIP-156 For VIP-157, have four more items. Field Description Drop Prefix The rule of add or replace code. If setup as Disable, it will add the prefix number prior to the identification number. If setup as Enable, it will replace the identification number. Prefix The prefix number. It only accepts the numeral and the max length is 8. Rule Rule The identification number. It can accept the numeral or symbol and the max length is 40.
SIP Analog Telephone Adapter - x: It is equal to 0~9. For example, [5xx] even if the number begins with 5. Dial Now rule If the dialing number matches this field, it will dial out and need not have to press the “#” key to end the dialing. It accepts the numeral or symbol, and the max length are 124. The starting number can’t be the “0”. For example, if the number is “0xxxx”, because the starting number is “0”, so that the system will ignore this dial plan.
SIP Analog Telephone Adapter Example_2: Drop prefix: Enable, Prefix: 006, Rule: 002+003+004+005+007+009 1. If the dialing number is “002+86xxxx”, it will match the rule [002], then system will automatically replace the prefix [002] to the prefix number [006].The real dialing number is [006+8613xxxxx]. 2. If the dialing number is “003+77xxxx”, it will match the rule [003], then system will automatically replace the prefix [003] to the prefix number [006]. The real dialing number is [006+77xxxx].
SIP Analog Telephone Adapter To enable this function, make sure that the PSTN line is connected to the PSTN port already. PSTN feature Code Default is 0*, the code for manually switching to PSTN line, and dial out from PSTN, it can only accept the numeric and *or #, the digital max length is 7. Routing Type Default is Disable, it defines the dialing route, according the [Routing Rule] to define the dialing route which is [IP or FXO]. Routing Rule Define the outgong rule.
SIP Analog Telephone Adapter VIP-157 Field Description Call Waiting Default is enable. When you are talking with other people, You can choose If you want to hear the notice when there is a new incoming call. If the call waiting function is On, if there is a new incoming call, you will hear the call waiting notice in your current call. If you set the function to Off, then you will not hear any notice. Ring Timeout Default is 60(sec). After how long the system will reply the busy(486 busy) message.
SIP Analog Telephone Adapter Auto Answer and PIN (VIP-157) Field Description Auto Answer Type Auto Answer: There are different incoming call types for flexible applications. The Trunk Gateway function needs to arrange with the registered Server System. The 3-Party subscribers could make Off-Net call (PSTN) through the FXO port of VIP-157. AutoAnswer Auto Answer Counter is to set after the ring count met the number you cournter set then the auto answer will enable.
SIP Analog Telephone Adapter Chapter 5 Network 5 Network Setting This page defines the LAN setting on this page. Field Description WAN Active The default is Fixed IP, and it also provides DHCP Client and PPPoE connection modes. Fixed IP: It could set up the IP address manually. DHCP Client: It will acquire the IP address automatically.
SIP Analog Telephone Adapter character, the maximum date length is 63. PPPoEService name PPPoE Service provider name PPPoE AC Name PPPoE AC name. DDNS Setting This page defines the DDNS setting on this page. You need to have the DDNS account and input the information properly. You can have a DDNS account with a public IP address, then others can call you via the DDNS account. But now most of the VoIP applications work with an SIP Proxy Server. When you finish the setting, please click the Submit button.
SIP Analog Telephone Adapter Field Description VLAN Activity If set up as On, it could receive VLAN messages. VID (802.1Q/TAG) Dispose VLAN ID is add a Tag header after enabling the VLAN function. The realized voice packets transfer is similar with that of VLAN. The prerequisite is it must be the same as VLAN of upper switch. The value range is 2~4094. User Priority To set up the user priority. (802.
SIP Analog Telephone Adapter Caution: VIP-156/VIP-157 VPN can’t use the encryption or compression for VPN connection. IPV6 Setting This page defines the IPV6 setting on this page. You can program the IPV6 information. Field Description IPV6 Activity Support three IPV6 types: Auto, Fixed IPV6, IPV6 in IPV4 Tunnel IPV6 address Setting the WAN IPV6 address or display it.(64 bits) SubnetPrefix Length Default is 64 Default Gateway IPV6 gateway address(64 bits) LAN IPv6 Address: IP V6 LAN address.
SIP Analog Telephone Adapter Chapter 6 NAT Trans 6 Stun Setting This page defines the STUN Enable/Disable and STUN Server IP address in this page. This function can help your Phone Adapter work properly behind NAT. To change these settings, please follow your ISP information. When you finish the setting, please click the Submit button. PC Setting This page defines the PC setting on this page. Field Description Device Activity The default is Bridge mode, and it also provides NAT mode.
SIP Analog Telephone Adapter address. PC IP address The IP address of PC port. (In the Birdge mode, the Default IP: 192.168.0.1 PC MAC Address The MAC of PC port Enable DHCP Server It will allot the IP address automatically when enable this function. IP Address The range for DHCP IP address. Lease Time DHCP server lease time DMZ and MAC Clone This page defines the DMZ and MAC Clone setting on this page.
SIP Analog Telephone Adapter Field Description Index The serial number. There are totally 12 records from Num 1 to 12. Activity The activity status. The default is Disable, this record will activate if enable. Protocol The TCP or UDP communication protocol. Internal Port For corresponding the internal port. External Port For corresponding the external port. Server IP To input the Server IP address.
SIP Analog Telephone Adapter Chapter 7 SIP Setting 7 Service Domain Setting In Service Domain Function, you need to input the account and the related information on this page. Please refer to your ISP provider. You can register five SIP accounts in the ATA. You can dial the VoIP phone to your friends via first enable SIP account and receive the phone from these five SIP accounts. Field Description Realm Which line you want to use.
SIP Analog Telephone Adapter system, it will frequently send the MWI message. The starting number can’t be the “0”. For example, if the number is “0xxxx”, because the starting number is “0”, so that the system will ignore this dial plan. You can see the Register Status on the Status page. If the item shows “Registered”, then your Phone Adapter is registered to the ISP, you can make a phone call direcly. If you have more than one SIP account, you can follow the steps below to register to the other ISP.
SIP Analog Telephone Adapter SIP Advance Setting This page defines the Hold by RFC, Voice/SIP QoS and other settings on this page. To change these settings, please follow your ISP information. When you finish the setting, please click the Submit button.
SIP Analog Telephone Adapter Field Description SIP Expire Time To set up the registration interval time. SIP Expire Time Default is General; Register interval time setting. Provide items like Type General (standard), 1/2, 2/3, 3/4, 4/5, 5/6, 6/7, 7/8, 8/9, 9/10。 Register server need supports this function Register time calculated General: expiry time-[(expiry time/30)*6], when Expiry Time>60 it will start to work, if less than 60 seconds, it will decrease 5 seconds. 1/2: expiry time * 1/2.
SIP Analog Telephone Adapter 5/6: expiry time * 5/6. 6/7: expiry time * 6/7. 7/8: expiry time * 7/8. 8/9: expiry time * 8/9. 9/10: expiry time * 9/10. SIP Register Retry If SIP register fails, system will retry interval after this time. Timer SIP session timer T1 Setting the maximum retransmit interval for non-INVITE requests and INVITE responses. Register server need supports this function SIP session timer T2 Setting the maximum retransmit interval for non-INVITE requests and INVITE responses.
SIP Analog Telephone Adapter Register server need supports this function Voice QoS The Voice QoS feature. (Diff-Serv) SIP QoS (Diff-Serv) The SIP QoS feature. The QoS setting is to set the voice packets’ priority. If you set the value higher than 0, then the voice packets will get the higher priority to the Internet. But the QoS function still needs to cooperate with the other Internet devices.
SIP Analog Telephone Adapter Send SIP PRACK to When sending the SIP package, in package Header will add Proxy: the ”PRACK” message。 Register server need supports this function Only Accept Trusted Only accept call from proxy, if system receives the IP dialing, system Certificates: will refuse the call.
SIP Analog Telephone Adapter Chapter 8 Advance Setting 8 Status Log Display and save systems running status message data. Press “Get Status Log” to back up the status log file. Auto Config This page defines the Auto Configuration (Auto Provision) setting. ATA supports TFTP, FTP, HTTP and IP PBX auto configuration function in total. In IP PBX Auto Configuration Setting you need to check with your service provider if they have provided this function.
SIP Analog Telephone Adapter Management-Advanced Setting This page defines the advanced functions. When you finish the setting, please click the Submit button.
SIP Analog Telephone Adapter Field Description ICMP Not Echo This function can disable echo when someone pings this device. It can avoid hacker trying to attack the device Anonymous Call If enable this function, machine will to start the calling hidden function, and it will not send the related Caller information. Register server need supports this function. Management form When [Enable] allow user login from WAN.
SIP Analog Telephone Adapter DHCP Gateway ARP The period to check the DHCP gateway ARP. Check Period Syslog Server IP There are seven Syslog types: Call Statistics, General Debug, Call Address Statistics + General Debug, SIP Debug, Call Statistics + SIP Debug, General Debug + SIP Debug and All. System Log Machine could send the system logs to the specific Syslog Server.
SIP Analog Telephone Adapter TR-069 On this page you can program the TR-069 setting. Different TR-069 server may need to modify some different parameters. What’s TR-069: Technical Report 069 (TR-069) is a customer-premises equipment WAN management protocol (CWMP) technical specification for remote management of end-user devices introduced by the broadband forum (formerly the DSL forum).TR-069 is an integrated framework equipped with safe auto-configuration. It also can take control of other CPE functions .
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SIP Analog Telephone Adapter Chapter 9 Other Setting 9 System Authority In System Authority it can change admin/System/User login password. Firmware Upgrade This page defines the SIP and RTP port number on this page. Each ISP provider will have different SIP/RTPport settings, please refer to the ISP to set up the port number correctly. When you finish the setting, please click the Submit button. If your update file is xxxx.ROM. you must enter http://VIP-15X’s-IP Address/update.htm e.g.http://192.168.0.
SIP Analog Telephone Adapter For technological consideration, we strongly suggest you refer to the following upgrade methods for updating your device. After firmware is loaded, the unit will reboot, and Default IP address of the customized firmware: http://192.168.0.1; login name/password: root/null (no password) Auto Update Setting The device can update new firmware with the gz or ds file format automatically by the Auto Upgrade function.
SIP Analog Telephone Adapter time. - Scheduling: The machine will follow the scheduled date and time to check the new firmware. Scheduling (Date) The machine will check the new firmware between the time range by random. Automatic Update There are Notify only and Automatic ways to update. - Notify only: If there are new firmware, the ATA will send the “Be Be Be” sound when picking up the handset to prompt there is new firmware. - Automatic: The device will carry out firmware update automatically.
SIP Analog Telephone Adapter Reset to default In Default Setting you can restore the Phone Adapter to factory default on this page. You can just click the Restore button, then the Phone Adapter will restore to default and automatically restart again. Save and Reboot In Save & Reboot you can save the changes you have done. If you want to use new setting in the Phone Adapter, you have to click the Save button.
SIP Analog Telephone Adapter Appendix A Voice Communication Samples There are several ways to make calls to desired destination in ATA. In this section, we’ll lead you step by step to establish your first voice communication via keypad and web browsers operations. Case 1: ATA to ATA connection via IP address Assume there are two ATAs in the network; the IP addresses are 192.168.0.1, 192.168.0.2 Analog telephone sets are connected to the phone (RJ-11) port of ATAs respectively 192.168.0.2 192.168.0.
SIP Analog Telephone Adapter Case 2: (Peer-to-Peer mode) VIP-157S Port 1 to Port 2 communications Supposing one VIP-157S connects to two telephones, just pick up phone 1 and dial ‘192*168*0*1**5062’, phone 2 will ring. Analog telephone sets are connected to the phone (RJ-11) ports of VIP-157S respectively 192.168.0.1 1001 1 9 2 * 1 1002 6 8 * 0 * 1 * * 5 0 6 2 # Test the scenario: 1. Pick up the telephone set on VIP-157S port 1, and you should be able to hear the dial-tone 2.
SIP Analog Telephone Adapter Machine configuration on the VIP-156: Please log in VIP-156_A via web browser and browse the Phone Settings menu and select the Call service config menu. On the setting page, please enable the All Forward function and fill in the Forward Type and Forward Number of VIP-156_B, then the sample configuration screen is shown below: Test the scenario: 1. VIP-156_C picks up the telephone 2. Dial the number 1001(VIP-156_A), 3.
SIP Analog Telephone Adapter Machine configuration on the VIP-157: Please log in VIP-157_A via web browser and browse the Phone Settings menu and select the Call service config menu. On the setting page, please select the All Forward function to PSTN choice and fill in the Forward Type and Forward Number of PSTN Phone Number 11112222, then the sample configuration screen is shown below: Test the scenario: 1. VIP-156_C pick up the telephone 2. Dial the number 1001(VIP-157_A) 3.
SIP Analog Telephone Adapter Machine configuration on the VIP-157: Please log in VIP-157_A via web browser and browse the Phone Settings menu and select the Call service config menu. On the setting page, please select the All Forward function to IP choice and fill in the Forward Type and Forward Number of of VIP-156_B, and then the sample configuration screen is shown below: Test the scenario: 1. PSTN Phone Number 11112222 pick up the telephone 2. Dial the PSTN Phone Number 33334444(VIP-157_A) 3.
SIP Analog Telephone Adapter Machine configuration on the VIP-156: Please log in VIP-156_A via web browser and browse the Phone Settings menu and select the Call service config menu. On the setting page, please enable the All Forward function and fill in the Forward Type and Forward Number of VIP-156_B, and then the sample configuration screen is shown below: Test the scenario: 1. VIP-156_C pick up the telephone 2. Dial the IP Address 192.168.0.1(VIP-156_A) 3.
SIP Analog Telephone Adapter Machine configuration on the VIP-157: STEP 1: Please log in VIP-157_A via web browser and browse the Phone Settings menu and select the Call service config menu. On the setting page, please disable All Forward function, and then the sample configuration screen is shown below: STEP 2: Please log in VIP-157_A via web browser and browse the Phone Settings / General setting menu and select the Auto Answer config menu.
SIP Analog Telephone Adapter 6. The Phone Number 11112222 will ring up; then it picks up the telephone and communication with the VIP-156_C Case 8: Auto Answer Feature_PSTN to IP For this example, there are one VIP-157 and two VIP-156 and connect with Peer to Peer mode. The VIP-157_A has set Auto Answer function for forwarding to arbitrary telephone.
SIP Analog Telephone Adapter Test the scenario: 1. The Phone Number 11112222 picks up the telephone 2. Dial the PSTN Phone Number 33334444(VIP-157_A) 3. VIP-157_A will ring up but doesn’t answer the call 4. After 3 rings, the Phone Number 11112222 will hear the prompt sounds and then input the password 123# 5. The Phone Number 11112222 will hear the dial tone and then input 0# 6. The IP address 192.168.0.
SIP Analog Telephone Adapter Appendix B The method of operation guide In this section, we’ll introduce the steps of how to set up some call features of the ATA. Please follow the steps below to utilize those features. Call Transfer A. Blind Transfer 1. B call to A and they are in the process of conversation. 2. A carries out the transfer function (Press “transfer” button) to hold the conversation with B. 3. A presses “#510#” and hears the dial tone and then input the number of C (Followed by the “#” key).
SIP Analog Telephone Adapter Switch the Realm (Registration Proxy Server) ATA can register to three different SIP Proxies at the same time. It can receive any one of different SIP accounts incoming call, and it can switch to anyone’s SIP accounts for making calls through inputting the switch code. Realm switch code: 1*: Realm 1 2*: Realm 2 3*: Realm 3 4*: Realm 4 5*: Realm 5 For example, the default is realm 1, input the 2* (Followed by the # key) from keypad and hang up the telephone set.
SIP Analog Telephone Adapter Appendix C VIP-156/VIP-156PE/VIP-157/VIP-157S Specifications Product Model Hardware LAN PC FXS (for telephone set connection) FXO (PSTN connection) Protocols and Standard Standard Voice codec Fax support Voice Standard Protocols Network and Configuration Access Mode Management Dimensions (W x D x H) Operating Environment Power Requirements EMC/EMI SIP Analog Telephone Adapter VIP-156 VIP-156PE VIP-157 VIP-157S 1 x 10/100Mbps RJ-45 port (802.
SIP Analog Telephone Adapter Appendix D Planet DDNS Application Configuring PLANET DDNS steps: Step 1 Enable DDNS option through accessing web page of ATA device. Step 2 Select DDNS server provided, and register an account if you have not used yet. Let’s take dyndns.org as an example. Register an account at http://planetddns.
EC Declaration of Conformity For the following equipment: *Type of Product *Model Number : SIP Telephone Adapter : VIP-156 * Produced by: Manufacturer‘s Name : Manufacturer‘s Address: Planet Technology Corp. 11F, No 96, Min Chuan Road Hsin Tien, Taipei, Taiwan, R. O.C. is herewith confirmed to comply with the requirements set out in the Council Directive on the Approximation of the Laws of the Member States relating to 1999/5/EC R&TTE.
EC Declaration of Conformity For the following equipment: *Type of Product *Model Number : PoE SIP Telephone Adapter : VIP-156PE * Produced by: Manufacturer‘s Name : Manufacturer‘s Address: Planet Technology Corp. 11F, No 96, Min Chuan Road Hsin Tien, Taipei, Taiwan, R. O.C. is herewith confirmed to comply with the requirements set out in the Council Directive on the Approximation of the Laws of the Member States relating to 1999/5/EC R&TTE.
EC Declaration of Conformity For the following equipment: *Type of Product *Model Number : VoIP Analog Telephone Adapter (1*FXS + 1*FXO) : VIP-157 * Produced by: Manufacturer‘s Name : Manufacturer‘s Address: Planet Technology Corp. 11F, No 96, Min Chuan Road Hsin Tien, Taipei, Taiwan, R. O.C.
EC Declaration of Conformity For the following equipment: *Type of Product *Model Number : VoIP Analog Telephone Adapter (2*FXS) : VIP-157S * Produced by: Manufacturer‘s Name : Manufacturer‘s Address: Planet Technology Corp. 11F, No 96, Min Chuan Road Hsin Tien, Taipei, Taiwan, R. O.C.
