Planet Technology Network Card VIP 157S User Manual

VoIP Analog Telephone Adapter  
VIP-156/VIP156PE/VIP-157/VIP-157S  
User’s manual  
Version 3.1  
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Revision  
User’s Manual for PLANET VoIP Analog Telephone Adapter:  
Model: ATA  
Rev: 3.1 (2008, February)  
Part No. EM-VIP_ATAV3.1  
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TABLE OF CONTENTS  
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Chapter 1  
1
Introduction  
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. The service  
providers and enterprises offer users traditional and enhanced the telephony communication services  
via the existing broadband connection to the Internet or corporation network.  
With the ATA, home users and companies are able to save the installation cost and extend their past  
investments in telephones, conference and speakerphones. The ATA can be the bridge between  
traditional analog systems and IP network with an extremely affordable investment.  
The ATA includes two alternatively Ethernet interface for Internet (PPPoE, DHCP or Fixed IP), or office  
LAN connection. With adding the auto-provision feature of our IP PBX product - IPX-2000, the ATA can  
be seamlessly integrated into the telephony network and be used in consumer and business IP  
telephony service, no expertise required!  
The ATA and our IP PBX system integration are the ideal combination for your office daily  
communications.  
Product Features  
Feature-rich telephone service over home or office Internet/Intranet connection  
Auto-config feature for flexible, ease-of-use IP PBX system integration  
Cost effective, field proven compatibility, and stability  
Web-based and telephone keypad machine configuration  
Remote machine administration authentication  
Voice prompt for machine configurations  
VoIP Features  
SIP 2.0 (RFC3261) compliant  
Peer-to-Peer / SIP proxy calls  
Voice codec support: G.711, G.723.1, G.729A/G.729B  
T.38 FAX transmission over IP network  
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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.  
Respective model/descriptions are shown below:  
VIP-156: SIP Analog Telephone Adapter  
VIP-156PE: 802.3af PoE SIP Analog Telephone Adapter  
VIP-157: 1 FXS/ 1 FXO SIP Analog Telephone Adapter  
VIP-157S: 2-port FXS SIP Analog Telephone Adapter  
Front Panel of VIP-156  
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Left / Right Panel of VIP-156  
Front Panel of VIP-156PE  
Left / Right Panel of VIP-156PE  
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Front Panel of VIP-157  
Left / Right Panel of VIP-157  
Front Panel of VIP-157S  
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Left / Right Panel of VIP-157S  
LED Display & Button  
RJ-45 connector, to maintain the existing network structure,  
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2
PC  
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.  
LAN  
The LAN interface also can be connected with 802.3af PoE  
switch or converter for power supply (VIP-156PE)  
12V DC Power input outlet  
3
4
12V DC  
Reset  
Reset to the factory default setting  
Machine default IP is http://192.168.0.1. Press RESET  
buttononrearpanelover5secondswillresettheVoIPPhone  
Adapter to factory default value. (Except speed dial and  
call forward settings)  
ÍNote  
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LED display of VIP-156 / VIP-156PE  
LED Indicators  
Descriptions  
Power is supplied to the device.  
PWR  
The Status LED will be flashing when the machine is operational  
STATUS  
LNK/ACT  
OFF: the device is connected to LAN at 10Mb/s.  
ON: the device is connected to LAN at 100Mb/s.  
OFF: the phone is idle.  
RING  
ON: the phone is in use (offhook).  
Blinking: the phone is ringing.  
LED display of VIP-157 / VIP-157S  
LED Indicators  
Descriptions  
The Status LED will be flashing when the machine is operational  
STATUS  
LNK/ACT  
OFF: the device is connected to LAN at 10Mb/s.  
ON: the device is connected to LAN at 100Mb/s.  
OFF: the phone is idle.  
RING 1  
RING 2  
ON: the phone is in use (offhook).  
Blinking: the phone is ringing.  
OFF: the phone is idle.  
ON: the phone is in use (offhook).  
Blinking: the phone is ringing.  
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Chapter 2  
2
Preparations & Installation  
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.  
For Internet Access, an Internet Access account with an ISP, and either of a DSL or Cable modem  
Administration Interface  
PLANET ATA provides GUI (Web based, Graphical User Interface) for machine management and  
administration.  
Web configuration access  
To start ATA web configuration, you must have one of these web browsers installed on computer for  
management  
Microsoft Internet Explorer 6.0.0 or higher with Java support  
Default LAN interface IP address of ATA is 192.168.0.1. You may now open your web browser, and  
insert http://192.168.0.1 in the address bar of web browser to logon ATA web configuration page.  
ATA will prompt for logon username/password, please enter: root / null (no password) to continue  
machine administration.  
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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.  
ÍNote  
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 adddress bar.  
After logging on machine with username/password (default: root / no passwrd), browse to “Network”  
--> “LAN Settings” configuration menu:  
Parameter Description  
IP address  
LAN IP address of ATA  
Default: 192.168.0.1  
Subnet Mask  
Default Gateway  
LAN mask of ATA  
Default: 255.255.255.0  
Gateway of ATA  
Default: 192.168.0.254  
Network settings via Keypad commands  
The ATA series phone adapters support telephone keypad configurations, please connect analog  
telephone set and refer to the following table for machine network configurations.  
IVR Menu Choice  
#111#  
Machine operation  
Parameter(s)  
None  
Notes:  
Set DHCP client  
ATA will change to DHCP  
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Client  
Use the * (star) key  
DHCP will be disabled and  
#112xxx*xxx*xxx*  
xxx#  
Setup Static IP Address  
Set Network Mask  
when entering a decimal system will change to the  
point.  
Static IP type.  
Use the * (star) key  
#113xxx*xxx*xxx*  
xxx#  
when entering a decimal Must set Static IP first.  
point.  
Use the * (star) key  
#114xxx*xxx*xxx*  
xxx#  
Set Gateway IP Address  
Set Primary DNS Server  
when entering a decimal Must set Static IP first.  
point.  
Use the * (star) key  
#115xxx*xxx*xxx*  
xxx#  
when entering a decimal Must set Static IP first.  
point.  
Must unlock the protect  
function before set up  
Unlock  
None  
#190#  
network settings and ATA  
function via keypad.  
The system will be lock  
Lock  
None  
None  
and can’t set up network  
settings via keypad.  
The system will reboot  
automatically.  
#191#  
#195#  
Reboot  
The system will be reset to  
factory default value and  
reboot automatically.  
VIP-157 only  
#198#  
Factory Reset  
None  
None  
0  
To switch PSTN mode  
Following keypad commands can be used to display the network settings enabled on ATA via voice  
prompt.  
IVR Menu Choice  
Machine operation  
Notes:  
IVR will announce the current PC-port IP  
address of the ATA.  
Check PC IP Address  
#120#  
IVR will announce if DHCP in enabled or  
disabled.  
Check network connection Type  
Check the Phone Number  
#121#  
#122#  
IVR will announce current enabled VoIP  
number.  
IVR will announce the current network mask  
of the ATA.  
Check Network Mask  
#123#  
#124#  
Check Gateway IP Address  
IVR will announce the current gateway IP  
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address of the ATA.  
IVR will announce the current setting in the  
Primary DNS field.  
Check Primary DNS Server Setting  
Check LAN IP Address  
#125#  
#126#  
#128#  
IVR will announce the current LAN port IP  
address of the ATA.  
IVR will announce the version of the  
firmware running on the ATA.  
Check Firmware Version  
Following keypad commands can be used to set up the main function .  
IVR Menu Choice  
#138#  
Machine operation  
Enable call waiting  
Disable call waiting  
Update firmware  
Blind Transfer  
Parameter(s)  
Notes:  
Enable Call waiting  
Disable Call waiting  
Update firmware  
None  
None  
None  
#139#  
#160#  
ATA Bland Transfer  
ATA Attendant Transfer  
ATA 3-way calling  
#510#  
Attendant Transfer  
3-way calling  
#511#  
#512#  
ATA transfer IP call to  
PSTN side  
IP transfer to PSTN  
#514#  
1:G.711 u-Law, 2: G.711  
a-Law, 3: G.723.1, 4:  
G.729a, 5: G.726 16K, 6:  
G.726 24K, 7: G.726  
32K, 8: G.726 40K,  
You can set the codec you  
want to the first priority.  
Set Codec  
#130+[1~8]#  
You can set the Handset  
gain to proper value,  
default is 10  
Handset Gain from  
0~15  
Set Handset Gain  
#131+[00~15]#  
#132+[00~12]#  
You can set the Handset  
volume to proper value,  
default is 10  
Handset Volume from  
0~12  
Set Handset Volume  
Set Auto config TFTP  
Server IP Address  
Set Auto config FTP  
Server IP Address  
You can set the TFTP  
Server IP address  
#135xxx*xxx*xxx*  
xxx#  
TFTP Server IP Address  
FTP Server IP Address  
You can set the FTP  
Server IP address  
#136xxx*xxx*xxx*  
xxx#  
You can set the Auto  
configuration mode, 0:  
Disable, 1: use TFTP  
Server, 2: user FTP Server  
0: Disable, 1: TFTP  
mode, 2: FTP mode  
Auto config mode  
#137+[0~2]#  
Forward function disable  
enable forward to FXS  
Disable forwrad funciton  
Eanble forward to FXS  
#145#  
#146+Number#  
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Port  
Port  
enable forward to FXO  
Port  
Eanble forward to FXO  
#147+Number#  
Port  
Enable PPTP function  
Disable PPTP function  
Enable VLAN function  
Disable VLAN function  
None  
None  
None  
None  
Enable PPTP function  
Disable PPTP function  
Enable VLAN function  
Disable VLAN function  
#116#  
#117#  
#118#  
#119#  
PleasecontactyourInternetserviceprovidertoobtainthe  
Internetaccesstype,andselectthepropernetworksettings  
in ATA to establish the network connections.  
L Hint  
After confirming the modification you’ve done, Please click on the Submit button to apply settings and  
browse to “Save & Reboot” menu to reboot the machine to make the settings effective.  
Connection Type  
Data required.  
In most circumstances, it is no need to configure the DHCP settings.  
The ISP will assign IP Address, and related information.  
Fixed IP  
DHCP clinet  
The ISP will assign PPPoE username / password for Internet access,  
PPPoE  
PleaseconsultyourISPpersonneltoobtainproperPPPoE/IP  
address related information, and input carefully.  
If Internet connection cannot be established, please check  
the physical connection or contact the ISP service staff  
for support information.  
L Hint  
Save Modification to Flash Memory  
Most of the VoIP router parameters will take effective after modifications, but it is just temporary stored  
on RAM only, it will disappear after your reboot or power off the VoIP Phone Adapter, to save the  
parameters into Flash ROM and let it take effective forever, please remember to press the Save &  
Reboot button after you modify the parameters.  
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Chapter 3  
Network Service Configurations  
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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:  
Š More comprehensive setting feels than traditional command line interface.  
Š Provides user input data fields, check boxes, and for changing machine configuration settings  
Š Displays machine running configuration  
To start ATA web configuration, you must have one of these web browsers installed on computer for  
management  
Š Microsoft Internet Explorer 6.0.0 or higher with Java support  
Manipulation of ATA via web browser  
Log on ATA via web browser  
After TCP/IP configurations on your PC, you may now open your web browser, and input  
http://192.168.0.1 to logon Phone Adapter web configuration page.  
Phone Adapter will prompt for logon username/password: root / null (no password)  
ATA log in page  
When users login the web page, users can see the Phone Adapter system information like firmware  
version, company…etc in this main page.  
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VoIP Phone Adatper main page  
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Chapter 4  
4
VoIP Telephone Adapter Configurations  
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  
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.  
Phone Book Page  
Phone  
Name  
The name of Phone Book records, it only can input  
numerals.  
Fill in the outgoing number (Line Number) or IP  
address.  
URL  
To select this record.  
Select  
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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. User pick up the handset and dial the “301” to make  
the P2P call (301@192.168.1.2).  
Ex_2:  
Users pick up the handset and dial the “206” to make the Proxy call (17476433364).  
Ex_3:  
Users pick up the handset and dial the “202” to make the P2P call (192.168.1.2:5062).  
Call Forward  
This page defines Call Forward function. You can setup the phone number you want to forward in this  
page. There are three type of Forward mode. You can choose All Forward, Busy Forward, and No  
Answer Forward by click the icon.  
All Forward: All incoming call will forward to the number you chosen. You can input the name and the  
phone number in URL field. If you select this function, then all the incoming call will direct forward to the  
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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.  
When you finished the setting, please click the Submit button.  
Call Forward function for VIP-156/VIP-156PT/VIP-157S  
Call Forward to PSTN (VIP-157): VIP-157 not only support 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 URL/Number field.  
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Call Forward function for VIP-157  
SNTP settings  
This page defines the primary and second SNTP Server IP Address, to get the date/time information.  
Also you can base on your location to set the Time Zone, and how long need to synchronize again.  
When you finished the setting, please click the Submit button.  
Volume Setting  
This page defines the Handset Volume, Ringer Volume, and the Handset Gain. When you finished the  
setting, please click the Submit button.  
Handset Volume is to set the volume for you can hear from the handset.  
Handset Gain is to set the volume send out to the other side’s handset.  
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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.  
Block Period: Set a time period and the phone will be blocked during the time period. If the “From”  
time is large than the “To” time, the Block time will from Day 1 to Day 2.  
When you finished the setting, please click the Submit button.  
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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 Counter is to set after the ring count met the number you set then the auto answer will  
enable.  
For security issue, You’d better to set the PIN Code. If you have set the PIN code, you will hear a tone  
to inform you input the PIN Code then you can dial out.  
Caller ID settings  
This page defines the device to show Caller ID in your PSTN Phone or IP Phone. There are four  
selection of Caller ID. You need to base on your environment to set the Caller ID function for FSK or  
DTMF.  
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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  
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.  
Drop Prefix  
The prefix number. It only accept the numeral and the max length is 8.  
The identification number. It can accept the numeral or symbol and  
Replace rule  
+
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the max length is 40.  
-
-
Symbol: It only accept the [+], [x]  
+: It means as “or”. For example, [123+456+334+5xx] even if  
[123 or 456 or 334 or 5xx]  
-
x: It is equal to 0~9. For example, [5xx] even if the number  
begin 5.  
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.  
Dial Now  
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.  
Stop dialing after seconds then send dial number out.  
If setup as Yes, the symtem sill stop to receive the dialing number  
when receive the [#] key. The system also will to determine the Auto  
Dial Time, it will carry out the calling if there isn’t receive the digit after  
the Auto Dial Time.  
Auto Dial Time  
Use # as send key  
If setup as No, the system just according to the Auto Dial Time to  
determine the end time.  
If setup as Yes, the system will look on [*] as [.]. For example, if dial  
the “192*168*0*100#”, it will dial out as “192.168.0.100#”.  
If setup as No, it just look on [*] as [*]. For example, if dial the “700*#”,  
it will dial out as “700*#”.  
Use * for IP dialing  
Descriptions of example:  
Example_1: Drop prefix: No, Replace rule 1: 002, +: 1234+4321 (No limit the digit length)  
1. If the dialing number is start as 1234, it will add the 002 at begin. The real dialing number is  
[0021234].  
2. If the dialing number is start as 4321, it will add the 002 at begin. The real dialing number is  
[0024321].  
Example_2: Drop prefix: Yes, Replace rule 2: 006, +: 002+003+004 (No limit the digit length)  
1. If the dialing number is start as 002, it will replace 002 by 006. The real dialing number is  
[006].  
2. If the dialing number is start as 003, it will replace 003 by 006. The real dialing number is  
[003].  
Example_3: Drop prefix: No, Replace rule 3: 007, +: 5xxx+35xx (Has limit the digit length)  
1. If the dialing number start as 5and follow 3 digits, it will add the 007 at begin. The real  
dialing number is [0075xxx].  
2. If the dialing number start as 35and follow 2 digits, it will add the 007 at begin. The real  
dialing number is [00735xx].  
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Example_4: 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/ 113119.  
4. If the dialing number is match with the rule of 8 digits, it will send out the dialing number  
directly. For example, 12345678.  
Flash Time Setting  
When you use the PSTN Phone and you need to press the Hook 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.  
Flash Time Settings for VIP-156/VIP-156PT/VIP-157S  
Beside the above settings, VIP-157 also can set the flash time of FXO port.  
Flash Time Settings for VIP-157  
Call waiting Settings  
When you are talking with other people, You can choose If you want to hear the notice when there is a  
new coming call. If the call waiting function is On, if there is a new incomeing call, you will hear the call  
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waiting notice in your current call. If you set the function to Off, then you will not hear any notice.  
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  
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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.  
LAN Settings  
This page defines the LAN setting in this page.  
LAN Mode: The default is Bridge mode, and it also provides NAT mode.  
-
-
Bridge: When set as is mode, the LAN and PC ports are in the same network segment.  
NAT: The LAN and PC ports are in the different network segment, and PC port could enable  
the DHCP Server function to allot the IP address.  
IP Type: The default is Fixed IP, and it also provides DHCP Client and PPPoE connection modes.  
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-
-
-
Fixed IP: It could setup the IP address manual.  
DHCP Client: It will acquire the IP address automatically.  
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.  
Password: The PPPoE connection account password. It could inpout numeral or character, the  
maximum date length are 63.  
PC Settings  
This page defines the PC setting in this page.  
IP: The IP address of PC port. (In the Birdge mode, the Default IP: 192.168.0.1)  
Mask: The sub net address. (Default: 255.255.255.0)  
MAC: The MAC of PC port  
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DHCP Server: It will allot the IP address automatically when enabke this function.  
Start IP: Start IP of lease table  
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. When  
you finished the setting, please click the Submit button.  
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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.  
CFI: To indicate the Canonical Format.  
-
-
If CFI=1, it means the header label include RIF field, and the NCIF flag valus of RIF will to  
decide the MAC address is Canonical Format or Non-Canonical Format in frame information.  
If CFI=0, it means the header label does not include RIF field, and the MAC address is  
Canonical Format in frame information.  
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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.  
Num: The serial number. There are total 24 records from Num 0 to 23.  
Enable: The activate status. The default is Disable, this record will been activate if enable.  
Protocol: The TCP or UDP communication protocol.  
Internal Port: For corresponding the internal port.  
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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.  
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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.  
Line number: you need to input the User Name get from your ISP.  
Register Name: you need to input the Register Name get from your ISP.  
Register Password: you need to input the Register Password get from your ISP.  
Domain Server: you need to input the Domain Server get from your ISP.  
Proxy Server: you need to input the Proxy Server get from your ISP.  
Outbound Proxy: you need to input the Outbound Proxy get from your ISP. If your ISP does not  
provide the information, then you can skip this item.  
You can see the Register Status in the Status item. 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 following the steps to register to the other ISP.  
When you finished the setting, please click the Submit button.  
LNote:  
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 any one SIP accounts for making calls through  
input the switch code.  
Realm switch code:  
1*: Realm 1  
2*: Realm 2  
3*: Realm 3  
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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 Settings  
This page defines the Codec priority, RTP packet length, and VAD function in this page. You need to  
follow the ISP suggestion to setup these items. When you finished the setting, please click the Submit  
button.  
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Codec ID Setting  
This page defines the Codec ID. Sometimes 2 VoIP devices 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.  
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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.  
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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.  
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 need to cooperate with the  
others Internet devices.  
SIP Expire Time: To setup the registration interval time.  
Use DNS SRV: The default is disable, and use DNS SRV mode. Set enable to use DNS to SRV mode  
to search the host information.  
STUN settings  
This page defines the STUN Enable/Disable and STUN Server IP address in this page. This function  
can help your Phone Adapter working properly behind NAT. To change these settings please following  
your ISP information. When you finished the setting, please click the Submit button.  
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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.  
PTT Settings  
In PTT Settings is for you to set the Country, different country will have different settings in FXS inter  
face.  
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PTT Settings for VIP-156/VIP-157S  
Beside the above settings, VIP-157 also can set country of FXO port.  
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. This function will copy the MAC address from NIC  
(Network Interface Card) which placed in PC to LAN port of ATA. That because some ISP will limit the  
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MAC address for PPPoE dial-up connection.  
Please refer to the following operate procedures for more understandings to carry out the MAC Clone  
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. (In the NAT mode, 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. ATA will prompt if sure want to clone the MAC of your PC to the LAN port of ATA.  
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5. After Save&Reboot, the MAC of LAN port will become to PC’s original MAC address.  
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.  
Send Anonymous CID: If enable this function, machine will to start the calling hidden function, and it  
will not send the related Caller information. (The Registration Server also need  
support this function)  
Billing Signal: There are provide three type billing types: Polarity Reversal, Tone_12K and Tone_16K.  
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(The Registration Server also need support this function)  
CPC Delay: When receive the disconnect signal, machine will to cut the voltage down to 0V after this  
time.  
CPC Duration: When starting to cut the voltage down to 0V, machine will to continue this state by this  
time.  
Send Flash event: There are provide two flash formats: DTMF Event and SIP Info.  
SIP Encrypt: There are provide seven encrypt formats: INFINET, AVS, WALKERSUN1,  
WALKERSUN2, CSF1, CSF2 and GX. (The Registration Server also need support this  
function)  
PPPoE retry period: If PPPoE dial-up connection fail, machine will retry the dial-up motion after this  
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.  
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Save & 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. After you click the Save button, the Phone Adapter  
will automatically restart and the new setting will effect.  
Firmware Upgrade  
In Firmware Upgrade function you can update new firmware via HTTP or TFTP methods in this page.  
You can ugrade the firmware by the following steps:  
Select the upgrade method and the firmware code type, 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.  
Select the correct file you want to download to the device then click the Update button.  
LNote:  
For technological consideration, we’ve strongly suggested refering to the following upgrade  
methods for update your device.  
After firmware loaded, the unit will be reboot, and Default IP address of the customized firmware:  
http://192.168.0.1; login name/password: root/null (no password)  
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Auto Upgrade  
The device can update new firmware with the gz or ds file format automatically by the Auto Upgrade  
function.  
Field  
Descriptions  
There are TFTP/ FTP and HTTP three ways to provide the auto  
Update via  
upgrade function.  
Input the TFTP Server address, and it could input the IP or Domain  
TFTP Server  
HTTP Server  
Name form.  
Input the HTTP Server address, and it could input the IP or Domain  
Name form.  
Set up the file path.  
HTTP File Path  
FTP Server  
Input the FTP Server address, and it could input the IP or Domain  
Name form.  
The login username.  
FTP Username  
FTP Password  
FTP File Path  
The login password  
Set up the file path.  
The device will according to the below ways to check the new  
Check new firmware  
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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.  
The machine will check the new firmware between the time range  
by random.  
Scheduling (Date)  
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” sounds when pick up the handset to prompt  
there are new firmware.  
- Automatic: The device will carry firmware update out  
automatically.  
It will check the information of model name.  
It will show the next check date and time.  
Firmware File Prefix  
Next update time  
LNote:  
If the Check new firmware field selected to Power On, the machine will chck the new  
firmware accoeding the scheduling time/date and power on. If there are new firmware can  
be upgraded, the machine won’t carry firmware update out automatic. The machine will send  
the prompt sounds when pick up the handset, and it needs to update firmware by manual.  
Reset to Default  
In Default Setting you can restore the Phone Adapter to factory default in this page. You can just click  
the Restore button, then the Phone Adapter will restore to default and automatically restart again.  
Reboot without saving  
Reboot function you can restart the Phone Adapter. If you want to restart the Phone Adapter, you can  
just click the Reboot button, then the Phone Adapter will automatically.  
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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  
192.168.0.1  
192.168.0.2  
1
9
2
*
1
6
8
*
0
*
2
#
Test the scenario:  
1. Pick up the telephone set on ATA A.  
2. Press the keypad: 192*168*0*2# shall be able to connect to the ATA B.  
3. Then the phone in 192.168.0.2 should ring. Please repeat the same dialing steps on ATA B to  
establish the first voice communication from ATA A  
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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  
1002  
1
9
2
*
1
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. Press the keypad: 192*168*0*1**5062# shall be able to connect to the VIP-157S port 2  
3. Then the telephone set in VIP-157S port 2 should ring. Please repeat the same dialing steps on  
port 2 to establish the first voice communication from VIP-157S  
y If the IP address of the remote calling party is known,  
you may directly make calls via its IP address and end  
with a “#”.  
L Hint  
y If the ATAs are installed behind a NAT/firewall/IP  
sharing device for Peer-to-Peer VoIP application,  
please make sure the NAT device support SIP  
applications, and suitable settings should be applied  
to the NAT device to enable the SIP communications  
before making calls  
y
[VIP-157S] in PLANET ATA series products, to connect  
to remote ATA, press the keypad in the following  
sequence to connect to the remote VIP-157S port 2:  
[remote ATA IP address]**5062, for example:  
192*168*0*2**5062  
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Case 3: Voice communication via SIP proxy server – SIP-50  
Registration /  
Authentication  
Registration /  
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 2:  
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 insert the account/password information  
obtained from your service provider (in this sample, we’re using PLANET SIP-50 as the SIP  
Proxy server for SIP account, call authentications), and then the sample configuration  
screen is shown below:  
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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. Please repeat the same dialing steps on ATA B to  
establish the first voice communication from ATA A  
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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: 192.168.0.2  
Line Number: 2002  
Device configurations on the IPX-2000:  
STEP 1:  
Log in IPX-2000 and browse to the Srevice Æ DHCP Service menu and create new options  
list for the auto configuration.  
Code: please insert 151 as the DHCP server option.  
An example option 151 would be option=151 value= http://192.168.0.50/tftpboot/  
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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.  
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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.  
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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.  
STEP 8:  
Repeat the same configuration steps on ATA B, and check if the ATA B is successfully  
registered with the IPX-2000 as one of the IP extensions.  
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. Please repeat the same dialing steps on ATA B to  
establish the first voice communication from ATA A  
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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. In the setting page, please enable the All Forward function  
and fill in the Name and URL of VIP-156_B, then the sample configuration screen is shown  
below:  
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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. The number 2002(VIP-156_B) will ring up then it pick up the telephone and communication  
with the number 3003(VIP-156_C)  
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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. In the setting page, please select the All Forward function to  
PSTN choice and fill in the Name and URL/Number of PSTN Phone Number 11112222,  
then the sample configuration screen is shown below:  
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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. The PSTN Phone Number 11112222 will ring up then it pick up the telephone and  
communication with the number 3003(VIP-156_C)  
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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. In the setting page, please select the All Forward function to  
IP choice and fill in the Name and URL/Number of of VIP-156_B, and then the sample  
configuration screen is shown below:  
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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. The number 2002(VIP-156_B) will ring up then it pick up the telephone and communication  
with the PSTN Phone Number 11112222  
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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. VIP-156_C pick up the telephone  
2. Dial the IP Address 192.168.0.1(VIP-156_A)  
3. Because VIP-156_A had set up All Forward function to the IP Address 192.168.0.2  
(VIP-156_B)  
4. The IP Address 192.168.0.2 (VIP-156_B) will ring up then it pick up the telephone and  
communication with the VIP-156_C  
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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.  
Machine configuration on the VIP-157:  
STEP 1:  
Please log in VIP-157_A via web browser, browse to the Phone Settings menu and select  
the Call Forward config menu. In 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, browse to the Phone Settings menu and select  
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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. The Phone Number 11112222 will ring up then it pick up the telephone and communication with  
the VIP-156_C  
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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.  
Machine configuration on the VIP-157:  
STEP 1:  
Please log in VIP-157_A via web browser, browse to the Phone Settings menu and select  
the Auto Answer config menu. In the setting page, please enable the Auto Answer and  
PIN Code Enabled function, and then the sample configuration screen is shown below:  
STEP 2:  
Please log in VIP-157_A via web browser, browse to the Phone Book menu and select the  
Speed Dial Settings config menu. In the setting page, please add a speed dial number for  
dial to IP address 192.168.0.2 (VIP-156_B), and then the sample configuration screen is  
shown below:  
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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 192.168.0.2 (VIP-156_B) will ring up then it pick up the telephone and  
communication with the Phone Number 11112222  
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Appendix B The method of operation guide  
In this section, we’ll introduce the features method of operation, and lead you step by step to establish  
these 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. C will ring up and A will get the busy tone for prompting to hang up  
5. C picks up the handset and conversation with B.  
B. Attendant Transfer  
1. B call to A and they are in the process of conversation.  
2. A carry the transfer function out to hold the conversation with B.  
3. A press “#511#” and hear the dial tone, then input the number of C (Follow by the “#” key).  
4. C will ring up.  
5. C picks up the handset and conversation with A.  
6. A hang up and C conversation with B.  
3-Way Conference  
1. A and B are in the process of conversation.  
2. A want to invite C to join their conversation.  
3. A press “Transfer” or “Hold” button to hold the conversation with B at first, then press “#512#”  
and hear the dial tone, then input the number of C (plus the “#” key).  
4. C will ring up and pick up the handset to conversation with A.  
5. A press “Transfer” button again, and they will entry the 3-Way conference mode.  
Call Waiting  
1. A and B are in the process of conversation.  
2. C call to A and A will hear the prompt sounds.  
3. A press “Hold” button to hold the conversation with B, and switch to conversation with C.  
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 any one SIP accounts for making calls through input  
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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.  
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Appendix C VIP-156/VIP-156PE/VIP-157/VIP-157S Specifications  
Product  
SIP Analog Telephone Adapter  
VIP-156 VIP-156PE  
Model  
VIP-157  
VIP-157S  
Hardware  
LAN  
1 x 10/100Mbps RJ-45 port (802.3af PoE for VIP-156PE)  
1 x 10/100Mbps RJ-45 port  
PC  
FXS (for telephone set  
connection)  
FXO (PSTN connection)  
Protocols and Standard  
Standard  
1 x RJ-11  
2 x RJ-11  
---  
---  
SIP 2.0 (RFC3261)  
1 x RJ-11  
Voice codec  
Fax support  
Voice Standard  
G.711a/u, G.723.1 (6.3k/5.3k), G.726, G.729A, G.729B, GSM  
T.38  
Voice activity detection (VAD)  
Comfort noise generation (CNG)  
Acoustic echo canceller (AEC)  
G.165: Line echo canceller (LEC)  
Jitter Buffer  
Protocols  
SIP 2.0 (RFC-3261), TCP//IP, UDP/RTP/RTCP, HTTP, ICMP, ARP, DNS,  
DHCP, NTP/SNTP, PPP, PPPoE  
Network and Configuration  
Access Mode  
Static IP, PPPoE, DHCP  
Web, keypad  
Management  
Dimension (W x D x H)  
Operating Environment  
Power Requirement  
EMC/EMI  
94 x 72 x 30 mm  
0~40 degree C, 10~95% humidity  
12V DC  
CE, FCC Class B  
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EC Declaration of Conformity  
For the following equipment:  
*Type of Product  
*Model Number  
:
:
SIP Telephone Adapter  
VIP-156  
* Produced by:  
Manufacturer‘s Name  
:
Planet Technology Corp.  
Manufacturer‘s Address:  
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.  
For the evaluation regarding the R&TTE, the following standards were applied:  
Emission  
Conducted / Radiated EN 55022  
(1998 + A1:2000 Class B)  
(1995 Class A)  
(1995)  
(1998 + A1:2001)  
(1995)  
(1995)  
(1995)  
(1995)  
(1996)  
Harmonic  
Flicker  
Immunity  
ESD  
EN 61000-3-2  
EN 61000-3-3  
EN 55024  
EN 61000-4-2  
EN 61000-4-3  
EN 61000-4-4  
EN 61000-4-5  
EN 61000-4-6  
EN 61000-4-8  
EN 61000-4-11  
EN 60950 3rd  
RS  
EFT/ Burst  
Surge Test  
CS  
Magnetic Field  
Voltage Disp  
Safety  
(1993)  
(1994)  
(2000)  
Responsible for marking this declaration if the:  
Manufacturer  
Authorized representative established within the EU  
Authorized representative established within the EU (if applicable):  
Company Name:  
Planet Technology Corp.  
Company Address: 11F, No.96, Min Chuan Road, Hsin Tien, Taipei, Taiwan, R.O.C  
Person responsible for making this declaration  
Name, Surname  
Position / Title :  
Jimmy Lin  
Product Manager  
Taiwan  
Place  
7th July, 2005  
Date  
Legal Singnature  
PLANET TECHNOLOGY CORPORATION  
e-mail: sales@planet.com.tw  
11F, No. 96, Min Chuan Road, Hsin Tien, Taipei, Taiwan, R.O.C. Tel:886-2-2219-9518 Fax:886-2-2219-9528  
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