Planet TechnologyメーカーVIP-157の使用説明書/サービス説明書
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VoIP Analog Telephone Adapter VIP-156 VIP-157 User’s manual 1.
Copyright Copyright (C) 2006 PLANET T echnology Corp. All rights reserved. The products and programs described in this User ’s Manu al are licensed products of PLANET T echnology , This User ’s Ma.
3 Revision User ’s Manual for PLANET V oIP Analog T elephone Adapter: Model: A T A Rev: 1.0 (2006, January) Part No. EM-VIP_A T A V1.
4 T able of Contents Chapter 1 Intr oduction ............................................................................................................ 6 Overview ......................................................................................
5 Firmware Upgrade .............................................................................................................................. 3 2 Reset to Default ....................................................................................
1 Chapter 1 Introduction Overview Based on years of VoIP manufacturing experien ces , PLANET Technology VoIP total solutio ns are known for advanced implementation of standards bas ed telephony with mass deployment capability.
7 • Voice processing: Voice Active Detection, DT MF detection/ generat ion, G.168 echo cancellation (16mSec.), Comfort noise gene ration (CNG) • In band, out-of-band, and SIP-info DTMF support Pac.
Physical Det ails The following figure illustrates the front/rear panel of ATA. Respective model/descriptions a re sho wn below: VIP-156: SIP Analog Telephone Adapter VIP-156PE: 802.
Front Panel of VIP-156PE Left / Right Panel of VIP-156PE Front Panel of VIP-157 9.
Left / Right Panel of VIP-157 Front Panel of VIP-157S Left / Right Panel of VIP-157S 10.
LED Display & Button 1 PC RJ-45 connector, to maintain t he existi ng network structure, connected directly to the PC through straight CAT-5 cable 2 LAN RJ-45 connector, for Internet a ccess, connected directly to Sw it ch /H ub through straight CAT-5 cable.
LED display of VIP-157 / VIP-157S LED Indicators Descriptions ST A TUS The S t atus LED will be flashing when the machine i s operational LNK/ACT OFF : the device is connected to LAN at 10Mb/s. ON : the device is connected to LAN at 100Mb/s. RING 1 OFF : the phone is idle.
2 Chapter 2 Preparations & Installation Physical Inst allation Requirement This chapter illustrates ba sic installation of ATA analog Phone Adapter ((“ ATA ” in the following term)) • Network cables. Use standard 10/1 00Ba seT network (UTP) cables with RJ45 connectors.
Í Note 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.
15 Network settings via Keypad commands The A T A series phone adapters supp ort telephone keyp ad configurations, please con ne ct analog telephone set and refer to the following t able for machine network configurations.
#123# Check Network Mask IVR will announce the current net work mask of the A T A. #124# Check Gateway IP Address IVR will announce the current gate way IP address of the A T A. #125# Check Primary DNS Server Setting IVR will announce the curre nt setting in the Primary DNS field.
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3 Chapter 3 Network Service Configurations Configuring and monitoring your Phone Adapter from web browser The A T A integrates a web-based graphi cal user in terface that can cover most configurations and machine st atus monitoring. V ia standard web browse r , you can configure and ch eck ma chine status from anywhere around the world.
When users login the web p age, users can see the Phone Ada pter system information like firmware version, company…etc in this main page. V oIP Phone Adatper main page 19.
4 Chapter 4 VoIP Telephone Adapter Configurations Speed Dial settings In S peed Dial setting functi on you ca n add/delete S peed Dial number . Y ou can input maximum 10 entries speed dial list. If you need to add a phone number into the S peed Dial list, you need to input the position, the name, and the phone number (by URL type).
Call forward Call Forward function: you can setup the phone number you want to forward in this p age. There are three type of Forward mode. Y ou can choose All Forw ard, Busy Forward, and No Answer Forward by click the icon. All Forward : All incoming call will forward to the n um ber you ch oo sed.
[VIP-156] Volume Setting This page defines the Han dset V olume, Ringer V olume, and the Handset Gain. When you finished the setting, please click the Submit button. Handset V olume is to set the volume for you can hear from the handset. Ringer V olume is to set the ringer volume for you can hear .
Volume Setting of VIP-157 Block Setting This page d efines the Block Setting to keep the ph one slience. Y ou can choose Always Block or Block a period. Always Block : All incoming call will be bloc ked until disable this feature. Block Period : Set a time period and the phone will be blocke d duri ng the time period.
after the ring count meet the numbe r yo u set then the auto answer will enable. Caller ID settings This page defines the d evice to sho w Caller ID in your PSTN Phone or IP Phone. There are fou r selection of Caller ID. Y ou need to base on your envir onment to set the Caller ID function for FSK or DTMF .
Flash Time Setting When you use the PSTN Phone and you need to press the Ho ok to do the Flash (Switch to the other phone line or HOLD), this function is for you to set the time you press the Hook to represent the Flash function.
Port Settings This page defines the SIP and RTP port number in th is p a ge. Each ISP provider will have dif ferent SIP/RTPport setting, please refer to the ISP to set up the port number correctly . When you finished the setting, please click the Submit button.
27 Service Domain Settings In Service Domain Function you need to input the a ccount and the related informations in thi s p age, please refer to your ISP provider . Y ou can register three SIP account in the A T A. Y ou can dial the V oIP phone to your friends via first enable SIP account and receive the phone from these three SIP accounts.
Codec Settings This page defines the Cod ec priority , RTP packet length, and V AD function in this page. Y ou need to follow the ISP suggestion to setup these items.
RTP Setting This page defines the O ut-Band DTM F and Send DTMF SIP Info Enable/Disa ble in this page. T o change this setting, please following your ISP informat ion. When you finished the setting, please click the Submit button. DTMF Setting This page def ines the RFC2833 Out-Band DTMF , Inband DTMF and Send DTMF SIP Info in this p age.
RPort Settings This page defines the RPort Enable/Disa ble in this page. T o 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, V oice/SIP QoS and SIP expire time in this p a ge.
your ISP information. Whe n you finishe d t he setting, please cli c k the Submit bu tton. Auto Configuration In Auto Configuratio n Setting you need to check with your service provider if they have provided this function. Usually this functi on will be boundle with an IP PBX to use in the of fice.
System Authority In System Authority you can cha nge you r login password. Save & Reboot In Save & Reboot you can save the changes you have done.
Click the “Browse” button in the ri ght side of the File Location or you can type th e correct path and the filename in File Location blank. Select the correct file you want to download to the Phone Adapter then click the Update button. Reset to Default In Default Setting you can restore the Phone Ada pter t o factory default in this page.
Appendix A V oice Communication Samples There are several ways to make call s to desired dest ination in A T A. In this section, we’ll lead you step by step to establish your first voice com muni cation via keyp ad and web browsers operations. ATA to ATA connection via IP address Assume there are two A T As in the network the IP address are 192.
(Peer-to-Peer mode) VIP-157S Port 1 to Port 2 communications Supposing one VIP-157S connect s to two telephones, just pick up phone 1 and di al ‘ 192*168*0*1**5062 ’, phon e 2 will ring. Analog telephone sets a re connected to the phon e (RJ-1 1) ports of VIP-157S respectively 192.
Voice communication via SI P proxy server – SIP-50 ATA A IP Address: 192.168.0.1 Line Number: 1001 ATA B IP Address: 192.168.0.2 Line Number: 2002 Registration / Authentication Registration / Authentication SIP-50 IP Address: 192.
37 STEP 3: Repeat the same configuration ste ps on A T A B , and che ck the machine registration st atus, make sure the regi stratio ns are completed. T est the scen ario: T o verify the V oIP communi.
Voice communication via IP PBX system – IPX-2000 (Auto-config) In the following sample, we’ll introduce how to in tegrate the A T A with our IP PBX system IPX-2000 via Auto-config feature. Registration / Authentication Registration / Authentication IPX-2000 LAN IP Address: 192.
STEP 3: Please press the Show extensions button to create the two extension ac counts/password: 1001/123 (for A T A A ), and 1002/1 23 (for A T A B) for the voice calls.
Device configurations on the A T A: STEP 5: Please log in A T A via web browser , bro wse to the SIP setting menu and select the Domain Service config menu. In the setting page, please bro wse to the Auto-config page, and enable the Auto Configurati on f eatures for IP PBX system.
41 Appendix B VIP-156 / VIP-157 S pecifications Product SIP Telephone Adapter Model VIP-156 VIP-156 PE VIP-157 VIP-157S Hardware LAN 1 x 10/100Mbps RJ-45 port (802.3af PoE for VIP-156PE) PC 1 x 10/100Mbps RJ-45 port FXS (for telephone set connection) 1 x RJ-11 2 x RJ-11 FXO (PSTN connection) --- 1 x RJ-11 --- Protocols and Standard Standard SIP 2.
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