VoIP Analog Telephone Adapter VIP-156/VIP156PE/VIP-157/VIP-157S User’s manual Version 3.
Copyright Copyright (C) 2010 PLANET Technology Corp. All rights reserved. The products and programs described in this User’s Manual are licensed products of PLANET Technology, This User’s Manual contains proprietary information protected by copyright, and this User’s Manual and all accompanying hardware, software, and documentation are copyrighted.
Revision User’s Manual for PLANET VoIP Analog Telephone Adapter: Model: VIP-156 / VIP156PE / VIP-157 / VIP-157S Rev: 3.31 (2010, October) Part No. EM-VIP_ATAV3.
TABLE OF CONTENTS Chapter 1 Introduction ............................................................................................................ 6 Overview....................................................................................................................................................... 6 Package Content .......................................................................................................................................... 7 Physical Details .......................
Codec ID Setting ................................................................................................................................ 38 DTMF Setting .................................................................................................................................... 39 RPort Settings..................................................................................................................................... 39 Other Settings.................................................
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, easy-to-install and simple-to-use, the PLANET VIP-156/VIP-157/VIP-157S VoIP Phone Adapter (“ATA” in the following term) converts standard telephones to IP-based networks.
• Voice processing: Voice Active Detection, DTMF detection/ generation, G.168 echo cancellation (16mSec.), Comfort noise generation (CNG) • In band, out-of-band, and SIP-info DTMF support Package Content The contents of your product should contain the following items: VoIP Telephone Adapter Power adapter Quick Installation Guide User’s Manual CD RJ-11 cable x 1 Physical Details The following figure illustrates the front/rear panel of ATA.
Left / Right Panel of VIP-156 Front Panel of VIP-156PE Left / Right Panel of VIP-156PE 8
Front Panel of VIP-157 Left / Right Panel of VIP-157 Front Panel of VIP-157S 9
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.
LED display of VIP-156 / VIP-156PE LED Indicators Descriptions PWR Power is supplied to the device. STATUS The Status LED will be flashing when the machine is operational LNK/ACT OFF: the device is connected to LAN at 10Mb/s. ON: the device is connected to LAN at 100Mb/s. RING OFF: the phone is idle. ON: the phone is in use (offhook). Blinking: the phone is ringing.
Chapter 2 Preparations & Installation 2 Physical Installation Requirement 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.
ÍNote 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 P in the adddress bar. Execute your web browser, and insert the IP address (default: 192.168.0.
Client #112xxx*xxx*xxx* xxx# #113xxx*xxx*xxx* xxx# #114xxx*xxx*xxx* xxx# #115xxx*xxx*xxx* xxx# Setup Static IP Address 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 Set Network Mask when entering a decimal Must set Static IP first. point. Use the * (star) key Set Gateway IP Address when entering a decimal Must set Static IP first. point.
address of the ATA. #125# Check Primary DNS Server Setting #126# Check LAN IP Address #128# Check Firmware Version 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. Following keypad commands can be used to set up the main function .
Chapter 2 Preparations & Installation 2 Physical Installation Requirement 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.
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. Overview on the web interface of ATA With web graphical user interface, you may have: More comprehensive setting feels than traditional command line interface.
VoIP Phone Adatper main page 18
Chapter 4 VoIP Telephone Adapter Configurations 4 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 Phone Name URL Select Description The default is Page 1. It can select Page1 ~ Page 14 to look round Phone Book records. The record number from 0 ~ 139, it can set up 140 records in total. The name of Phone Book records, it only can input numerals. Fill in the outgoing number (Line Number) or IP address.
If you need to add a phone number into the Phone Book list, you need to input the position, the name, and the phone number (by URL type). When you finished a new phone list, just click the “Add Phone” button. If you want to delete a phone number, you can select the phone number you want to delete then click “Delete Selected” button. If you want to delete all phone numbers, you can click “Delete All” button. For Example: Ex_1: ATA had added the above phone numbers.
speed dial number you choose. Busy Forward: If you are on the phone, the new incoming call will forward to the number you choosed. You can input the name and the phone number in URL field. No Answer Forward: If you can not answer the phone, the incoming call will forward to the number you chosen. You can input the name and the phone number in URL field. Also you have to set the Time Out time for system to start to forward the call to the number you choosed.
The IP Line Forward function is use for the incoming call is IP call type, and the destination is IP or PSTN call types. The FXO Line Forward function is use for the incoming call is PSTN call type, and the destination is IP call type. The IP / FXO Line Forward functions can be functioned at the same time, and that could separate different incoming call types for flexable applications.
Volume Settings for VIP-154T/VIP-154PT Beside the above settings, VIP-157 also can set the volume of PSTN. PSTN-Out Volume is to set the PSTN volume for you can hear. PSTN-In Gain is to set the volume send out to the other side’s handset. Volume Settings for VIP-157 Block Setting This page defines the Block Setting to keep the phone slience. You can choose Always Block or Block a period. Always Block: All incoming call will be blocked until disable this feature.
Auto Answer (For VIP-157) This page defines the Auto Answer function. You can set the Auto Answer function to answer the incoming call by the phone. If the call is come from the IP, then the VIP-157 can let user to redial the call to PSTN phone number. If the call is coming from PSTN, then the VIP-157 can let user to redial to IP Phone number. Auto Answer: There are different incoming call types for flexable applications. The Trunk Gateway function needs to arrange in with the registered Server System.
Dial Plan Settings 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.
Field Description Routing to To select use IP or FXO types for auto routing function, and according to the Routing rule to handle the dialing numbers. Routing rule The rule can delete the prefix number. Drop Prefix The rule of add or replace code. If setup as No, it will add the prefix number prior to the identification number. If setup as Yes, it will replace the identification number. Replace rule The prefix number. It only accept the numeral and the max length is 8.
[123 or 456 or 334 or 5xx] - x: It is equal to 0~9. For example, [5xx] even if the number begin 5. Dial Now If the dialing number are match with this field, it will dial out and need not to press the “#” key to end the dialing. It accepts the numeral or symbol, and the max length are 124. LNote: 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.
Example_5: Dial Now: *xx+#xx+11x+xxxxxx 1. If the dialing number is match with the rule of “*xx”, it will send out the dialing number directly. For example, *00/ *01/ *02…*99. 2. If the dialing number is match with the rule of “#xx”, it will send out the dialing number directly. For example, #00/ #01/ #02…#99. 3. If the dialing number is match with the rule of “11x”, it will send out the dialing number directly. For example, 111/ 112/ 113…119. 4.
T.38 (FAX) Settings This page defines the T.38 (FAX) Setting function. You can Enable/Disable the T.38 function, and can modify the T.38 transmission port of each FXS port. T.38 (FAX) Settings for VIP-156/VIP-156PE/VIP-157 T.38 (FAX) Settings for VIP-157S Hot line Settings This page defines the Hot line setting in this page. When user pick up the handset, the device will call to the specific number automatically. Use Hot Line: Click Enable to carry the Hot line function out.
Hot line number: The hot line number, it can input the IP address or registration number. Alarm Settings This page defines the Alarm setting in this page. It provides the alarm function, and it can set up the Alarm Time to get the telephone ringed up every day. Alarm: The default is Off. If set up as On, the telephone will ringed up at the specific time. Alarm Time: It can set up the system prompt time with 24 hours. Current time: The next alarm time.
- PPPoE: It will use the PPPoE connection method. IP: The IP address Mask: The sub net address Gateway: The default gateway address DNS Server1: The default is 168.95.192.1, it could setup the first DNS server address. DNS Server2: The default is 168.95.1.1, it could setup the second DNS server address. MAC: The MAC of LAN port Host Name: The product model User Name: The PPPoE connection account name. It could inpout numeral or character, the maximum date length are 63.
End IP: End IP of lease table. Network device connecting to the PC port can dynamic obtain the IP in the range between start IP and end IP Lease Time: DHCP server lease time DDNS Settings This page defines the DDNS setting in this page. You need to have the DDNS account and input the informations 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 are work with a SIP Proxy Server.
VLAN Settings This page defines the VLAN setting in this page. This function needs to co-operate with network devices which have VLAN function. VLAN Packets: If setup as On, it could receive VLAN messages. VID (802.1Q/TAG): Dispose VLAN ID is add a Tag header after realize enable the VLAN function. The realized voice packets transfer at the same VLAN. The prerequisite is it must the same as VLAN of upper switch. The value range are 2~4094. User Priority (802.1P): To setup the user priority.
DMZ Settings This page defines the DMZ setting in this page. DMZ: If setup as On, all of packets (expect SIP packets) will send to the specific IP address. DMZ Host IP: The DMZ host IP address. Virtual Server This page defines the Virtual Server setting in this page. You could define 24 virtual service information in this page. When you finished the setting, please click the Submit button. Virtual Server Page: There are total page1 to page 3. It could choose the page which want to go over.
External Port: For corresponding the external port. Server IP: To input the Server IP address. PPTP Settings This page defines the PPTP setting in this page. You could setup the PPTP Server connection information. When you finished the setting, please click the Submit button.
Service Domain Settings In Service Domain Function you need to input the account and the related informations in this page, please refer to your ISP provider. You can register three SIP account in the ATA. You can dial the VoIP phone to your friends via first enable SIP account and receive the phone from these three SIP accounts. First you need click Active to enable the Service Domain, then you can input the following items: Display Name: you can input the name you want to display.
For example: The default is realm 1, input the 2* (Follow by the # key) from keypad and hang up the telephone set. It will switch to realm 2, and it can make the SIP calls via realm 2. Port Settings This page defines the SIP and RTP port number in this page. Each ISP provider will have different SIP/RTPport setting, please refer to the ISP to setup the port number correctly. When you finished the setting, please click the Submit button.
Codec ID Setting This page defines the Codec ID. Sometimes 2 VoIP device with different Codec ID will cause the interopability issue. If you are talking with others got some problems, you may ask the other one what kind of Codec ID he use, then you can change your Codec ID. When you finished the setting, please click the Submit button.
DTMF Setting This page defines the RFC2833 Out-Band DTMF, Inband DTMF and Send DTMF SIP Info in this page. To change this setting, please following your ISP information. When you finished the setting, please click the Submit button. RPort Settings This page defines the RPort Enable/Disable in this page. To change this setting, please following your ISP information. When you finished the setting, please click the Submit button.
Other Settings This page defines the Hold by RFC, Voice/SIP QoS and other settings in this page. To change these settings please following your ISP information. When you finished the setting, please click the Submit button. Hold by RFC: The default is disable, and to start up communication hold back function (RFC definition). Set enable to start up the Hold by RFC function. Voice QoS (Diff-Serv): The Voice QoS feature. SIP QoS (Diff-Serv): The SIP QoS feature.
Auto Configuration 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. Usually this function will be boundle with an IP PBX to use in the office.
Encrypt Config This page can decide if use encryption key for Auto Configuration (Auto Provision) funciton. The encryption key needs be the same as config file for Auto Configuration. If the encryption key is wrong , ATA won’t to write in the settings when got the config file. If ATA cancel encryption key, it won’t need to input encryption key when generating the config file. PTT Settings In PTT Settings is for you to set the Country, different country will have different settings in FXS inter face.
PTT Settings for VIP-157 Cancel without to tag This function can decide the device if send the cancel tag to Proxy Server. If there has the similar symptom that the caller cancel the call before the called answer the call, the called still continue to ring up even the caller has cancel this call. It could try select this function to Yes to avoid the above symptom. MAC CIone Setting This page defines the MAC Clone Enable/Disable.
function. 1. Please login ATA and browse to “Network -> LAN Settings” page. To switch the LAN mode to NAT mode then press Save&Reboot button to save the settings and reboot machine. 2. Please make sure the network cable of your PC directly connect with PC port of ATA, then re-login ATA. (The default IP address of PC port is http://192.168.123.1 ) 3. Please browse to “Advanced Settings -> MAC Clone Setting” page and enable the MAC Clone function. 4.
Tone Settings This page defines the Tone settings. This function can setup the related parameters of Dial Tone, Ring Back Tone, Busy Tone, Error Tone and Inser Tone. When you finished the setting, please click the Submit button. Advanced Settings This page defines the advanced functions. When you finished the setting, please click the Submit button. ICMP Not Echo: This function can disable echo when someone ping this device, it can avoid haker try to attack the device.
time. System Log Server: Machine could send the system logs to the specific Syslog Server. It can input the IP or Domain address. System Log Type: There are seven Syslog types: Call Statistics, General Debug, Call Statistics + General Debug, SIP Debug, Call Statistics + SIP Debug, General Debug + SIP Debug and All. System Authority In System Authority you can change your login password. Save & Reboot In Save & Reboot you can save the changes you have done.
will automatically restart and the new setting will effect. Firmware Upgrade In Firmware Upgrade function you can update new firmware via HTTP method in this page. You can ugrade the firmware by the following steps: Select the upgrade method and the firmware code type, CPU (AP) or DSP code. Click the “Browse” button in the right side of the File Location or you can type the correct path and the filename in File Location blank.
Field Descriptions Update via There are TFTP/ FTP and HTTP three ways to provide the auto upgrade function. TFTP Server Input the TFTP Server address, and it could input the IP or Domain Name form. TFTP File Path Set up the file path. HTTP Server Input the HTTP Server address, and it could input the IP or Domain Name form. HTTP File Path Set up the file path. FTP Server Input the FTP Server address, and it could input the IP or Domain Name form. FTP Username The login username.
firmware. - Power On (+ Scheduling): The machine will check the new firmware when power on and following the scheduling date and time. - Scheduling: The machine will follow the scheduling 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.
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 address 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.
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.
Case 3: Voice communication via SIP proxy server – SIP-50 Registration / Registration / Authentication Authentication SIP-50 IP Address: 192.168.0.50 ATA A IP Address: 192.168.0.1 Line Number: 1001 ATA B IP Address: 192.168.0.2 Line Number: 2002 Device configurations on the ATA: STEP 1: Log in SIP-50 and create two testing accounts/password: 1001 / 123 (for ATA A), and 1002 / 123 (for ATA B) for the voice calls.
STEP 3: Repeat the same configuration steps on ATA B, and check the machine registration status, make sure the registrations are completed. Test the scenario: 1. Pick up the telephone on ATA A 2. Press the keypad: 2002 shall be able to connect to the ATA B 3. Then the telephone set in ATA B should ring.
Case 4: Voice communication via IP PBX system – IPX-2000 (Auto-config) In the following sample, we’ll introduce how to integrate the ATA with our IP PBX system IPX-2000 via Auto-config feature. Registration / Registration / Authentication Authentication IPX-2000 LAN IP Address: 192.168.0.50 ATA A IP Address: 192.168.0.1 Line Number: 1001 ATA B IP Address: 192.168.0.
STEP 2: Please browse to the Device Æ IP Phone menu and create new device. And press the EDIT button for set up the Auto Config configuration. STEP 3: Please fill out the Vendor Prefix code and MAC Address of ATA. LNote: The following are the Vendor Prefix of devices: 1. VIP-156: pla156 2. VIP-157/VIP-157S: pla157 STEP 4: Please browse to the Device Æ Extension of IP Phone menu to create the two extension accounts/password: 1001/123 (for ATA A), and 1002/123(for ATA B) for the voice calls.
STEP 5: After setting up the parameters, please browse to the Service Æ IP PBX service menu, and press RELOAD of IP PBX configuration reload selection for activating the settings. Device configurations on the ATA: STEP 6: Please log in ATA via web browser, browse to the SIP setting menu and select the Domain Service config menu. In the setting page, please browse to the Auto-config page, and enable the Auto Configuration features for IP PBX system.
STEP 7: After enabling the auto-config feature, the ATA shall be able to obtain IP address and SIP extension information from IP PBX system IPX-2000 information. To verify the auto-config results, you may connect telephone set to ATA; press #120# to check if the IP address is obtained from IPX-2000. And #122# can be used to verify the extension number assigned by IPX-2000.
Case 5: Call Forward Feature_Example 1 In the following samples, we’ll introduce the Call Forward Feature applications. In this example, there are three VIP-156 register to IPX-300 and VIP-156_A had set Call Forward function to VIP-156_B. Machine configuration on the VIP-156: Please log in VIP-156_A via web browser, browse to the Phone Settings menu and select the Call Forward config menu.
Test the scenario: 1. VIP-156_C pick up the telephone 2. Dial the number 1001(VIP-156_A), 3. Because VIP-156_A had set up All Forward function to the number 2002(VIP-156_B) 4.
Case 6: Call Forward Feature_Example 2 In this example, there are one VIP-157 and two VIP-156 register to IPX-300. The VIP-157_A had set Call Forward function to phone number 1111-2222 (PSTN). Machine configuration on the VIP-157: Please log in VIP-157_A via web browser, browse to the Phone Settings menu and select the Call Forward config menu.
Test the scenario: 1. VIP-156_C pick up the telephone 2. Dial the number 1001(VIP-157_A) 3. Because VIP-157_A had set up All Forward function to the PSTN Phone Number 11112222 4.
Case 7: Call Forward Feature_Example 3 In this example, there are one VIP-157 and two VIP-156 register to IPX-300. The VIP-157_A had set Call Forward function to number 2002 (VIP-156_B). Machine configuration on the VIP-157: Please log in VIP-157_A via web browser, browse to the Phone Settings menu and select the Call Forward config menu.
Test the scenario: 1. PSTN Phone Number 11112222 pick up the telephone 2. Dial the PSTN Phone Number 33334444(VIP-157_A) 3. Because VIP-157_A had set up All Forward function to the number 2002(VIP-156_B) 4.
Case 8: Call Forward Feature_Example 4 In this example, there are three VIP-156 and connect with Peer to Peer mode. VIP-156_A had set Call Forward function to VIP-156_B. Machine configuration on the VIP-156: Please log in VIP-156_A via web browser, browse to the Phone Settings menu and select the Call Forward config menu. In the setting page, please enable the All Forward function and fill in the Name and URL of VIP-156_B, and then the sample configuration screen is shown below: Test the scenario: 1.
Case 9: Auto Answer Feature_IP to PSTN In this example, there are one VIP-157 and two VIP-156 and connect with Peer to Peer mode. The VIP-157_A had set Auto Answer function for forwarding calls to arbitrary telephone. If there have incoming IP calls and VIP-157_A doesn’t answer the incoming calls after specific time, the caller will hear prompt sounds to input the password then dial out an arbitrary PSTN telephone.
the Auto Answer config menu. In the setting page, please enable the Auto Answer and PIN Code Enabled function, 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-157_A) 3. VIP-157_A will ring up but doesn’t answer the call 4. After 3 rings, the VIP-156_C will hear the prompt sounds then input the password 123# 5. VIP-156_C will hear the dial tone from PSTN line then input Phone Number 11112222 6.
Case 10: Auto Answer Feature_PSTN to IP In this example, there are one VIP-157 and two VIP-156 and connect with Peer to Peer mode. The VIP-157_A had set Auto Answer function for forwarding to arbitrary telephone. If there have incoming PSTN calls and VIP-157_A doesn’t answer the incoming calls after specific time, the caller will hear prompt sounds to input the password and then dial out an arbitrary IP telephone.
Test the scenario: 1. The Phone Number 11112222 pick 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 then input the password 123# 5. The Phone Number 11112222 will hear the dial tone then input 0# 6. The IP address 192.168.0.
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 carry the transfer function out (Press “transfer” button) to hold the conversation with B. 3. A press “#510#” and hear the dial tone, then input the number of C (Follow by the “#” key). 4.
the switch code. Realm switch code: 1*: Realm 1 2*: Realm 2 3*: Realm 3 For example: The default is realm 1, input the 2* (Follow by the # key) from keypad and hang up the telephone set. It will switch to realm 2, and it can make the SIP calls via realm 2. Auto Update firmware by manual (Keypad) If pick up the handsetof ATA, it will hear the “DoDoDo” prompt. If want to carry out the upgrade action, please input ”#190#” to unlock the device at first. Then input ”#160#” to upgrade the new firmware.
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 Dimension (W x D x H) Operating Environment Power Requirement EMC/EMI SIP Analog Telephone Adapter VIP-156 VIP-156PE VIP-157 VIP-157S 1 x 10/100Mbps RJ-45 port (802.
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.
