PLANET VIP-156PE User Manual

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VoIP Analog Telephone Adapter

VIP-156/VIP156PE/VIP-157/VIP-157S

User’s manual

Version 3.31

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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.

No part of this User’s Manual may be copied, photocopied, reproduced, translated, or reduced to any electronic

medium or machine-readable form by any means by electronic or mechanical. Including photocopying, recording,

or information storage and retrieval systems, for any purpose other than the purchaser's personal use, and without

the prior express written permission of PLANET Technology.

Disclaimer

PLANET Technology does not warrant that the hardware will work properly in all environments and applications,

and makes no warranty and representation, either implied or expressed, with respect to the quality, performance,

merchantability, or fitness for a particular purpose.

PLANET has made every effort to ensure that this User’s Manual is accurate; PLANET disclaims liability for any

inaccuracies or omissions that may have occurred.

Information in this User’s Manual is subject to change without notice and does not represent a commitment on the

part of PLANET. PLANET assumes no responsibility for any inaccuracies that may be contained in this User’s

Manual. PLANET makes no commitment to update or keep current the information in this User’s Manual, and

reserves the right to make improvements to this User’s Manual and/or to the products described in this User’s

Manual, at any time without notice.

If you find information in this manual that is incorrect, misleading, or incomplete, we would appreciate your

comments and suggestions.

CE mark Warning

The is a class B device, In a domestic environment, this product may cause radio interference, in which case the

user may be required to take adequate measures.

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To avoid the potential effects on the environment and human health as a result of the presence of

hazardous substances in electrical and electronic equipment, end users of electrical and electronic

equipment should understand the meaning of the crossed-out wheeled bin symbol. Do not dispose of

WEEE as unsorted municipal waste and have to collect such WEEE se

parately.

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The PLANET logo is a trademark of PLANET Technology. This documentation may refer to numerous hardware

and software products by their trade names. In most, if not all cases, their respective companies claim these

designations as trademarks or registered trademarks.

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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_ATAV3.31

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TABLE OF CONTENTS

Chapter 1 Introduction ............................................................................................................ 6

Overview....................................................................................................................................................... 6

Package Content .......................................................................................................................................... 7

Physical Details ............................................................................................................................................ 7

LED Display & Button....................................................................................................................... 10

Chapter 2 Preparations & Installation.................................................................................12

Physical Installation Requirement............................................................................................................ 12

LAN IP address configuration via web configuration interface .........................................................13

Chapter 3 Network Service Configurations......................................................................... 17

Configuring and monitoring your ATA from web browser.................................................................... 17

Overview on the web interface of ATA .............................................................................................. 17

Manipulation of ATA via web browser............................................................................................... 17

Chapter 4 VoIP Telephone Adapter Configurations............................................................ 19

Phone Book ........................................................................................................................................ 19

Call Forward....................................................................................................................................... 20

SNTP settings..................................................................................................................................... 22

Volume Setting ................................................................................................................................... 22

Block Setting ...................................................................................................................................... 23

Auto Answer (For VIP-157)............................................................................................................... 24

Caller ID settings................................................................................................................................24

Dial Plan Settings ............................................................................................................................... 25

Flash Time Setting.............................................................................................................................. 28

Call waiting Settings .......................................................................................................................... 28

T.38 (FAX) Settings............................................................................................................................ 29

Hot line Settings ................................................................................................................................. 29

Alarm Settings.................................................................................................................................... 30

LAN Settings...................................................................................................................................... 30

PC Settings ......................................................................................................................................... 31

DDNS Settings ................................................................................................................................... 32

VLAN Settings................................................................................................................................... 33

DMZ Settings ..................................................................................................................................... 34

Vir tua l Server ..................................................................................................................................... 34

PPTP Settings..................................................................................................................................... 35

Service Domain Settings ....................................................................................................................36

Port Settings ....................................................................................................................................... 37

Codec Settings.................................................................................................................................... 37

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Codec ID Setting ................................................................................................................................38

DTMF Setting .................................................................................................................................... 39

RPort Settings..................................................................................................................................... 39

Other Settings..................................................................................................................................... 40

STUN settings .................................................................................................................................... 40

Auto Configuration............................................................................................................................. 41

Encrypt Config ................................................................................................................................... 42

PTT Settings....................................................................................................................................... 42

Cancel without to tag.......................................................................................................................... 43

MAC CIone Setting............................................................................................................................ 43

Tone Settings ...................................................................................................................................... 45

Advanced Settings.............................................................................................................................. 45

System Authority................................................................................................................................46

Save & Reboot ...................................................................................................................................46

Firmware Upgrade.............................................................................................................................. 47

Auto Upgrade ..................................................................................................................................... 47

Reset to Default.................................................................................................................................. 49

Reboot without saving........................................................................................................................ 49

Appendix A Voice Communication Samples......................................................................................... 51

Case 1: ATA to ATA connection via IP address .................................................................................. 51

Case 2: (Peer-to-Peer mode) VIP-157S Port 1 to Port 2 communications ......................................... 52

Case 3: Voice communication via SIP proxy server – SIP-50 ............................................................ 53

Case 4: Voice communication via IP PBX system – IPX-2000 (Auto-config)................................... 55

Case 5: Call Forward Feature_Example 1.......................................................................................... 59

Case 6: Call Forward Feature_Example 2.......................................................................................... 61

Case 7: Call Forward Feature_Example 3.......................................................................................... 63

Case 8: Call Forward Feature_Example 4.......................................................................................... 65

Case 9: Auto Answer Feature_IP to PSTN......................................................................................... 66

Case 10: Auto Answer Feature_PSTN to IP....................................................................................... 68

Appendix B The method of operation guide............................................................................................ 70

Call Transfer....................................................................................................................................... 70

3-Way Conference .............................................................................................................................. 70

Call Waiting........................................................................................................................................ 70

Switch the Realm (Registration Proxy Server)................................................................................... 70

Auto Update firmware by manual (Keypad)...................................................................................... 71

Appendix C VIP-156/VIP-156PE/VIP-157/VIP-157S Specifications..................................................... 72

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Chapter 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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LED Display & Button

Left / Right Panel of VIP-157S

1 PC

2 LAN

3 12V DC

4 Reset

Machine default IP is http://192.168.0.1. Press RESET

Note

button on rear panel over 5 seconds will reset the VoIP Phone

dapter to factory default value. (Except speed dial and

call forward settings)

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.

The LAN interface also can be connected with 802.3af PoE

switch or converter for power supply (VIP-156PE)

12V DC Power input outlet

Reset to the factory default setting

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LED display of VIP-156 / VIP-156PE

ED Indicators DeL scriptions

PWR

STATUS

LNK/ACT

RING

Power is supplied to the device.

The Status LED will be flashing when the machine is operational

OFF: the device is connected to LAN at 10Mb/s.

ON: the device is conne

OFF: the phone is idle.

ON: the phone is in use (offhook

Blinking: the phone is ringing.

cted to LAN at 100Mb/s.

LED display of VIP-157 / VIP-157S

LED Indicators Descriptions

STATUS

LNK/ACT

RING 1

The Status LED will be flashing when the machine is operational

OFF: the device is connected to LAN at 10Mb/s.

ON: the device is conne

OFF: the phone is idle.

ON: the phone is in use (offhook

cted to LAN at 100Mb/s.

RING 2

Blinking: the phone is r

OFF: the phone is idle.

ON: the phone is in use (offhook

Blinking: the phone is ringing.

inging.

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Chapter 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

Note

(192.168.0.x) of ATA. If you’re not familiar with TCP/IP, please refer to related chapter on user’s manual CD or consult your network administrator for proper network configurations.

LAN IP address configuration via web configuration interface

Execute your web browser, and insert the IP address (default: 192.168.0.1) of VIP in the adddress bar.

After logging on machine with username/password (default: root / no passwrd), browse to “Network”

--> “LAN Settings” configuration menu:

P in the adddress bar.

Parameter Description

IP address LAN IP address of

Default: 192.168.0.1

Subnet Mask LAN mask of

Default: 255.255.255.0

Default Gateway Gateway of

Default: 192.168.0.254

ATA

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 Machine operation Parameter(s) Notes:

#111#

Set DHCP client None ATA will change to DHCP

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Client

#112xxx*xxx*xxx*

xxx#

#113xxx*xxx*xxx*

xxx#

#114xxx*xxx*xxx*

xxx#

#115xxx*xxx*xxx*

xxx#

#190#

Use the * (star) key

Setup Static IP Address

Set Network Mask

Set Gateway IP Address

Set Primary DNS Server

Unlock None

when entering a decimal

point.

Use the * (star) key

when entering a decimal

point.

Use the * (star) key

when entering a decimal

point.

Use the * (star) key

when entering a decimal

point.

DHCP will be disabled and

system will change to the

Static IP type.

Must set Static IP first.

Must unlock the protect

function before set up

network settings and ATA

function via keypad.

The system will be lock

#191#

#195#

#198#

0＊

Lock None

Reboot None

Factory Reset None

To switch PSTN mode None VIP-157 only

and can’t set up network

settings via keypad.

The system will reboot

automatically.

The system will be reset to

factory default value and

reboot automatically.

Following keypad commands can be used to display the network settings enabled on ATA via voice

prompt.

IVR Menu Choice Machine operation Notes:

#120#

#121#

Check PC IP Address

Check network connection Type

IVR will announce the current PC-port IP

address of the ATA.

IVR will announce if DHCP in enabled or

disabled.

#122#

#123#

#124#

Check the Phone Number

IVR will announce current enabled VoIP

number.

IVR will announce the current network mask

Check Network Mask

of the ATA.

Check Gateway IP Address IVR will announce the current gateway IP

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address of the ATA.

#125#

#126#

#128#

Check Primary DNS Server Setting

Check LAN IP Address

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 .

IVR Menu Choice Machine operation Parameter(s) Notes:

#138#

#139#

#160#

#510#

#511#

#512#

#514#

Enable call waiting None Enable Call waiting

Disable call waiting None Disable Call waiting

Update firmware None Update firmware

Blind Transfer ATA Bland Transfer

Attendant Transfer ATA Attendant Transfer

3-way calling ATA 3-way calling

IP transfer to PSTN

ATA transfer IP call to

PSTN side

#130+[1~8]#

#131+[00~15]#

#132+[00~12]#

#135xxx*xxx*xxx*

xxx#

#136xxx*xxx*xxx*

xxx#

Set Codec

Set Handset Gain

Set Handset Volume

TFTP Server IP Address

FTP Server IP Address

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,

Handset Gain from

0~15

Handset Volume from

0~12

Set Auto config TFTP

Server IP Address

Set Auto config FTP

Server IP Address

You can set the codec you

want to the first priority.

You can set the Handset

gain to proper value,

default is 10

You can set the Handset

volume to proper value,

default is 10

You can set the TFTP

Server IP address

You can set the FTP

Server IP address

You can set the Auto

#137+[0~2]#

#145#

#146+Number#

Auto config mode

0: Disable, 1: TFTP

mode, 2: FTP mode

Forward function disable Disable forwrad funciton

enable forward to FXS Eanble forward to FXS

configuration mode, 0:

Disable, 1: use TFTP

Server, 2: user FTP Server

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Chapter 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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Chapter 3

Network Service Configurations

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.

 Provides user input data fields, check boxes, and for changing machine configuration settings

 Displays machine running configuration

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

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

Phone Adapter will prompt for logon username/password: root / null (no password)

to logon Phone Adapter web configuration page.

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

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

Phone Book Page

Phone

Name

URL

Select

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.

To select this record.

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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.

IP Line Forward function for VIP-157

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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.

FXO Line 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 fo

Handset Gain is to set the volume send out to the other side’s handset.

r you can hear from the 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

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.

“From”

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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: There are different incoming call types for flexable applications. The Trunk Gateway

function needs to arrange in with the registered Server System. The 3-Party subscribers could make

Off-Net call (PSTN) through the FXO port of VIP-157.

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. Please notice that the PIN Code function

couldn’t function with Trunk Gateway function together.

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.

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Field

Routing to

Routing rule

Drop Prefix

Replace rule

Description

To select use IP or FXO types for auto routing function, and according

to the Routing rule to handle the dialing numbers.

The rule can delete the prefix number.

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.

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

the max length is 40.

- Symbol: It only accept the [+], [x]

- +: It means as “or”. For example, [123+456+334+5xx] even if

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[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

Realm 1/2/3 prefix

Auto Dial Time

Use # as send key

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.

These options can define the switching code for each Realm No.

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.

If setup as No, the system just according to the Auto Dial Time to

determine the end time.

Descriptions of example:

Example_1: Routing to: FXO, Routing rule: D007+009

1. If the dialing number is “00722199518”, it will delete 007 and send the following number for

dialing out via FXO port. The real dialing number is [22199518].

2. If the dialing number is “00922199518”, it won’t delect any number. The real dialing number is

[00922199518].

Example_2: Drop prefix: No, Replace rule 1: 002, +: 1234+4321 (No limit the digit length)

3. If the dialing number is start as “1234”, it will add the 002 at begin. The real dialing number is

[0021234…].

4. If the dialing number is start as “4321”, it will add the 002 at begin. The real dialing number is

[0024321…].

Example_3: 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_4: Drop prefix: No, Replace rule 3: 007, +: 5xxx+35xx (Has limit the digit length)

1. If the dialing number start as “5” and follow 3 digits, it will add the 007 at begin. The real

dialing number is [0075xxx].

2. If the dialing number start as “35” and follow 2 digits, it will add the 007 at begin. The real

dialing number is [00735xx].

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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. 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

waiting notice in your current call. If you set the function to Off, then you will not hear any notice.

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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.

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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.

- Fixed IP: It could setup the IP address manual.

- DHCP Client: It will acquire the IP address automatically.

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- 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. (Default: 192.168.0.1)

Mask: The sub net address. (Default: 255.255.255.0)

MAC: The MAC of PC port

DHCP Server: It will allot the IP address automatically when enabke this function.

Start IP: Start IP of lease table

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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.

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 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.

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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.

Send Keep Alive Packet: Always to transport the network packets to keep the NAT port could be

opened continuous.

Keep Alive Period: To setup the interval time for transporting packets.

Jitter Buffer: To setup the size for jitter buffer packets.

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.

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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-156/VIP-157S

Beside the above settings, VIP-157 also can set country of FXO port.

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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

MAC address for PPPoE dial-up connection.

Please refer to the following operate procedures for more understandings to carry out the MAC Clone

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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. ATA will prompt if sure want to clone the MAC of your PC to the LAN port of ATA.

5. After Save&Reboot, the MAC of LAN port will become to PC’s original MAC address.

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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.

(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

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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. 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

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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.

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)

Auto Upgrade

The device can update new firmware with the gz or ds file format automatically by the Auto Upgrade

function.

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Field Descriptions

Update via

TFTP Server

TFTP File Path

HTTP Server

HTTP File Path

FTP Server

FTP Username

FTP Password

FTP File Path

Check new firmware

There are TFTP/ FTP and HTTP three ways to provide the auto

upgrade function.

Input the TFTP Server address, and it could input the IP or Domain

Name form.

Set up the file path.

Input the HTTP Server address, and it could input the IP or Domain

Name form.

Set up the file path.

Input the FTP Server address, and it could input the IP or Domain

Name form.

The login username.

The login password

Set up the file path.

The device will according to the below ways to check the new

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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.

Scheduling (Date)

Automatic Update

Firmware File Prefix

Next update time

The machine will check the new firmware between the time range

by random.

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.

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

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

1 9 2

1001

* 1 6 8 * 1

1002

** 0 0 6 5 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

Hint

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 “#”.

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

 ATA A

IP Address: 192.168.0.1

Line Number: 1001

Device configurations on the ATA:

STEP 1:

STEP 2:

SIP-50 IP Address: 192.168.0.50

123 (for ATA B

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

) for the voice calls.

 ATA B IP Address: 192.168.0.2

Line Number: 2002

), and 1002 /

screen is shown below:

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STEP 3:

Repeat the same configuration steps on ATA B

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

establish the first voice communication from ATA A

should ring. Please repeat the same dialing steps on ATA B to

, and check the machine registration status,

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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 /

Authentication

Registration /

Authentication

IPX-2000 LAN IP Address: 192.168.0.50

 ATA A IP Address: 192.168.0.1  ATA B IP Address: 192.168.0.2

Line Number: 1001

Line Number: 2002

Device configurations on the IPX-2000:

STEP 1:

list for the auto configuration.

Code: please insert 151 as the DHCP server option.

Value: http://LAN IP for IPX-2000/tftpboot/

An example option 151 would be option=151 value= http://192.168.0.50/tftpboot/

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STEP 2:

STEP 3:

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.

Please fill out the Vendor Prefix code and MAC Address of ATA.

STEP 4:

LNote:

The following are the Vendor Prefix of devices:

1. VIP-156: pla156

2. VIP-157/VIP-157S: pla157

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

press RELOAD of IP PBX configuration reload selection for activating the settings.

Device configurations on the ATA:

STEP 6:

IP PBX service menu, and

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:

STEP 8:

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.

Repeat the same configuration steps on ATA B

registered with the IPX-2000 as one of the IP extensions.

, and check if the ATA B is successfully

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

establish the first voice communication from ATA A

should ring. Please repeat the same dialing steps on ATA B to

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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:

STEP 2:

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:

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 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. 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

Page 71

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 Model VIP-156 VIP-156PE 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) FXO (PSTN connection) --- 1 x RJ-11 --Protocols and Standard Standard SIP 2.0 (RFC3261) Voice codec G.711a/u, G.723.1 (6.3k/5.3k), G.726, G.729A, G.729B, GSM Fax support T.38 Voice Standard 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 Management Web, keypad Dimension (W x D x H) 94 x 72 x 30 mm Operating Environment 0~40 degree C, 10~95% humidity Power Requirement 12V DC EMC/EMI CE, FCC Class B

1 x RJ-11 2 x RJ-11

Page 73

EC Declaration of Conformity

For the following equipment:

*Type of Product

: SIP Telephone Adapter

*Model Number : 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) Harmonic EN 61000-3-2 (1995 Class A) Flicker EN 61000-3-3 (1995) Immunity EN 55024 (1998 + A1:2001) ESD EN 61000-4-2 (1995) RS EN 61000-4-3 (1995) EFT/ Burst EN 61000-4-4 (1995) Surge Test EN 61000-4-5 (1995) CS EN 61000-4-6 (1996) Magnetic Field EN 61000-4-8 (1993) Voltage Disp EN 61000-4-11 (1994) Safety EN 60950 3rd (2000)

Responsible for marking this declarati o n i f the:

⌧

Manufacturer  Authorized representative established within the EU Authorized representative established within the EU (if applicable): Company Name: Planet Techno l og y Cor p. Company Address: 11F, No.96, Min Chuan Road, Hsin Tien, Taipei, Ta iwan, R.O.C Person responsible for making this declaration Name, Surname Jimmy Lin Position / Title : Product Manager

Taiwan

7th July, 2005

Place Date Legal Singnature

PLANET TECHNOLOGY CORPORATION

e-mail: sales@planet.com.tw http://www.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

Page 74

EC Declaration of Conformity

For the following equipment:

*Type of Product

: PoE SIP Telephone Adapter

*Model Number : VIP-156PE

* Produced by: Manufacturer‘s Name : Planet Technology Corp. Manufacturer‘s Address: 11F, No 96, Min Chuan Road Hsin Tien, Taipei, Taiwan, R. O.C.

Responsible for marking this declarati o n i f the:

⌧

Taiwan

7th July, 2005

Place Date Legal Singnature

PLANET TECHNOLOGY CORPORATION

e-mail: sales@planet.com.tw http://www.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

Page 75

EC Declaration of Conformity

For the following equipment:

*Type of Product

: VoIP Analog Telephone Adapter (1*FXS + 1*FXO)

*Model Number : VIP-157

* 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 Electromagnetic Compatibility Directive on (89/336/EEC,92/31/EEC,93/68/EEC). For the evaluation regarding the EMC, the following standards were applied:

Conducted / Radiated EN 55022 (1998 + A1:2000 + A2:2003) Harmonic EN 61000-3-2 (2000) Flicker EN 61000-3-3 (1995 + A1:2001) Immunity EN 55024 (1998 + A1:2001) ESD EN 61000-4-2 (1995 + A1:1998 + A2:2000) RS EN 61000-4-3 (2002 + A1:2002) EFT/ Burst EN 61000-4-4 (1995 + A1:2000 + A2:2001) Surge Test EN 61000-4-5 (1995 + A1:2000) CS EN 61000-4-6 (1996 + A1:2000) Magnetic Field EN 61000-4-8 (1993 + A1:2000) Voltage Disp EN 61000-4-11 (1994 + A1:2000)

Responsible for marking this declarati o n i f the:

⌧

Name, Surname Jimmy Lin Position / Title : Product Manager

Taiwan

16 March, 2006

Place Date Legal Signature

PLANET TECHNOLOGY CORPORATION

e-mail: sales@planet.com.tw http://www.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

Page 76

EC Declaration of Conformity

For the following equipment:

*Type of Product

: VoIP Analog Telephone Adapter (2*FXS)

*Model Number : VIP-157S

* Produced by: Manufacturer‘s Name : Planet Technology Corp. Manufacturer‘s Address: 11F, No 96, Min Chuan Road Hsin Tien, Taipei, Taiwan, R. O.C.

Conducted / Radiated EN 55022 (1998 + A1:2000) Harmonic EN 61000-3-2 (2000) Flicker EN 61000-3-3 (1995 + A1:2001) Immunity EN 55024 (1998 + A1:2001) ESD EN 61000-4-2 (1995 + A1:2001 + A2:2000) RS EN 61000-4-3 (2002 + A1:2002) EFT/ Burst EN 61000-4-4 (1995 + A1:2000 + A2:2001) Surge Test EN 61000-4-5 (1995 + A1:2000) CS EN 61000-4-6 (1996 + A1:2000) Magnetic Field EN 61000-4-8 (1993 + A1:2000) Voltage Disp EN 61000-4-11 (1994 + A1:2000)

Responsible for marking this declarati o n i f the:

⌧

Name, Surname Jimmy Lin Position / Title : Product Manager

Taiwan

17 March, 2006

Place Date Legal Signature

PLANET TECHNOLOGY CORPORATION

e-mail: sales@planet.com.tw http://www.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