Planet VIP156PE User's Manual

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Users Manual of VIP-156/156PE/157S

VoIP Analog Telephone Adapter

VIP-156/VIP-156PE/VIP-157S


Copyright

Copyright© 2017 by PLANET Technology Corp. All rights reserved. No part of this publication may be reproduced, transmitted, transcribed, stored in a retrieval system, or translated into any language or computer language, in any form or by any means, electronic, mechanical, magnetic, optical, chemical, manual or otherwise, without the prior written permission of PLANET.

PLANET makes no representations or warranties, either expressed or implied, with respect to the contents hereof and specifically disclaims any warranties, merchantability or fitness for any particular purpose. Any software described in this manual is sold or licensed "as is". Should the programs prove defective following their purchase, the buyer (and not this company, its distributor, or its dealer) assumes the entire cost of all necessary servicing, repair, and any incidental or consequential damages resulting from any defect in the software. Further, this company reserves the right to revise this publication and to make changes from time to time in the contents hereof without obligation to notify any person of such revision or changes.

All brand and product names mentioned in this manual are trademarks and/or registered trademarks of their respective holders.

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.

Trademarks

The PLANET logo is a trademark of PLANET Technology Corp. This documentation may refer to numerous hardware and software products by their trade names. In most, if not all cases, these designations are claimed as trademarks or registered trademarks by their respective companies.

CE mark Warning

This 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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FCC Interference Statement


This equipment has been tested and found to comply with the limits for a Class B digital device, pursuant to

Part 15 of FCC Rules. These limits are designed to provide reasonable protection against harmful interference in a residential installation. This equipment generates, uses, and can radiate radio frequency energy and, if not installed and used in accordance with the instructions, may cause harmful interference to radio communications. However, there is no guarantee that interference will not occur in a particular installation. If this equipment does cause harmful interference to radio or television reception, which can be determined by turning the equipment off and on, the user is encouraged to try to correct the interference by one or more of the following measures:

1. Reorient or relocate the receiving antenna.

2. Increase the separation between the equipment and receiver.

3. Connect the equipment into an outlet on a circuit different from that to which the receiver is connected.

4. Consult the dealer or an experienced radio technician for help.

FCC Caution:

To assure continued compliance (example-use only shielded interface cables when connecting to computer or peripheral devices). Any changes or modifications not expressly approved by the party responsible for compliance could void the user’s authority to operate the equipment. This device complies with Part 15 of the

FCC Rules. Operation is subject to the Following two conditions: (1) This device may not cause harmful interference, and (2) this Device must accept any interference received, including interference that may cause undesired operation.

WEEE Caution

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 separately.

Safety

This equipment is designed with the utmost care for the safety of those who install and use it. However, special attention must be paid to the dangers of electric shock and static electricity when working with electrical equipment. All guidelines of this and of the computer manufacture must therefore be allowed at all times to ensure the safe use of the equipment.

Customer Service

For information on customer service and support for the Gigabit SSL VPN Security Router, please refer to the following Website URL: http://www.planet.com.tw

Before contacting customer service, please take a moment to gather the following information:

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 VoIP Analog Telephone Adapter serial number and MAC address

 Any error messages that displayed when the problem occurred

 Any software running when the problem occurred

 Steps you took to resolve the problem on your own


Revision

User’s Manual of PLANET VoIP Analog Telephone Adapter

Model: VIP-156/VIP-156PE/VIP-157S

Rev: 6.0 (October, 2017)

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TABLE OF CONTENTS

Chapter 1.

Introduction ...................................................................................................................... 7

1.1

Overview ........................................................................................................................... 7

1.2

Application...................................................................................................................... 11

1.3

Features .......................................................................................................................... 13

1.4

Package Contents .......................................................................................................... 14

1.5

Physical Details .............................................................................................................. 14

Chapter 2.

Preparations & Installation ........................................................................................... 18

2.1

Physical Installation Requirements ............................................................................. 18

2.2

LAN IP Address Configuration via Web Configuration Interface .............................. 19

Chapter 3.

Network Service Configurations .................................................................................. 23

3.1

Configuring and Monitoring your ATA from Web Browser ....................................... 23

3.1.1

Overview of the web interface of ATA ...................................................................... 23

3.1.2

Manipulation of ATA via web browser ...................................................................... 23

Chapter 4.

VoIP Telephone Adapter Configurations ..................................................................... 25

4.1

Status .............................................................................................................................. 25

4.2

Phone Book .................................................................................................................... 26

4.3

Call Service ..................................................................................................................... 28

4.4

SNTP setting ................................................................................................................... 30

4.5

Volume Setting ............................................................................................................... 30

4.6

Dial Plan Setting ............................................................................................................. 31

4.7

General ............................................................................................................................ 34

Chapter 5.

Network ........................................................................................................................... 35

5.1

Network Setting .............................................................................................................. 35

5.2

DDNS Setting .................................................................................................................. 36

5.3

VLAN Setting .................................................................................................................. 36

5.4

VPN Setting..................................................................................................................... 37

5.5

IPv6 Setting..................................................................................................................... 38

Chapter 6.

NAT Trans ....................................................................................................................... 39

6.1

Stun Setting .................................................................................................................... 39

6.2

PC Setting ....................................................................................................................... 39

6.3

DMZ and MAC Clone ...................................................................................................... 40

6.4

Virtual Server .................................................................................................................. 41

Chapter 7.

SIP Setting ...................................................................................................................... 42

7.1

Service Domain Setting ................................................................................................. 42

7.2

Codec Setting ................................................................................................................. 43

7.3

SIP Advance Setting ...................................................................................................... 44

Chapter 8.

Advance Setting ............................................................................................................. 48

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8.1

Status Log....................................................................................................................... 48

8.2

Auto Config..................................................................................................................... 49

8.3

Management -- Advanced Setting ................................................................................ 50

8.4

Tones ............................................................................................................................... 52

8.5

TR-069 ............................................................................................................................. 52

Chapter 9.

Other Setting .................................................................................................................. 54

9.1

System Authority ........................................................................................................... 54

9.2

Firmware Upgrade ......................................................................................................... 54

9.3

Auto Update Setting ...................................................................................................... 55

9.4

Reset to default .............................................................................................................. 57

9.5

Save and Reboot ............................................................................................................ 57

9.6

Logout ............................................................................................................................. 57

Appendix A Voice Communication Samples ................................................................................... 58

Case 1: ATA to ATA connection via IP address .................................................................... 58

Case 2: (Peer-to-Peer mode) VIP-157S Port 1 to Port 2 communications .......................... 58

Case 3: Call Forward Feature_Example 1 .............................................................................. 59

Case 4: Call Forward Feature_Example 2 .............................................................................. 60

Appendix B The method of operation guide .................................................................................... 62

Call Transfer .............................................................................................................................. 62

3-Way Conferencing ................................................................................................................. 62

Call Waiting ............................................................................................................................... 62

Switch the Realm (Registration Proxy Server) ...................................................................... 63

Auto Update Firmware Manually (Keypad) ............................................................................ 63

Appendix C VIP-156/VIP-156PE/ VIP-157S Specifications .............................................................. 64

Appendix D Planet DDNS Application .............................................................................................. 65

Configuring PLANET DDNS Steps: ......................................................................................... 65

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CHAPTER 1. INTRODUCTION

1.1 Overview

PLANET VoIP Adapter

PLANET VIP-156/VIP-156PE/VIP-157S, an easy-to-use-and-install Analog Telephone Adapter, provides the same voice quality and reliability that users have come to expect from PLANET ATAs.

The VIP-156/VIP-156PE/VIP-157S is a 1-/2-port Analog Telephone Adapter which allows you to use your existing analog phones or fax machines to make calls using the internet. The VIP-156/VIP-156PE/VIP-157S supports T.38 fax and works with most SIP-based internet service providers (ISPs). The

VIP-156/VIP-156PE/VIP-157S is a great choice for voice service providers because it is completely customized auto-provisioning and auto-update . If you want to have the world's leading VoIP service, you need to have the world's leading VoIP device!

Cost-effective, easy-to-install and simple-to-use, the VIP-156/VIP-156PE/VIP-157S converts standard telephones to IP-based networks. With the 802.3af/at PoE integration (VIP-156PE only), the service providers and enterprises can offer users traditional and enhanced the telephony communication services via the existing broadband connection to the Internet or corporation network.

Professional Application

The VIP-156/VIP-156PE/VIP-157S includes two Ethernet interfaces for Internet (PPPoE, DHCP or Fixed IP), or office LAN connection. Besides the IPv6 , VPN (PPTP and L2TP), VLAN and PLANET Easy Dynamic

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DNS features, the VIP-156/VIP-156PE/VIP-157S is an intelligent low-density Voice over IP (VoIP) gateway that enables carrier-class residential and business IP telephony services delivered over broadband or high-speed Internet connections to two standard RJ11 telephones. The 10/100 LAN port allows internet connectivity to be extended to a second device eliminating the need for a second Ethernet drop.

Calls from your VIP-156/VIP-156PE/VIP-157S to any other device on any VIP-156/VIP-156PE/VIP-157S device anywhere in the world are always free! The VIP-156/VIP-156PE/VIP-157S allows you to use your analog devices with any SIP trucking provider. Simply plug the Analog Telephone Adapter into your broadband router and then plug your analog devices into the ATA. The VIP-156/VIP-156PE/VIP-157S also can connect to your broadband router and to another IP device such as a PC. The 2-port RJ45 router has integrated QoS to ensure your voice traffic is prioritized above other types of traffic. It will support up to 5 VoIP services or SIP accounts .

When used with a SIP service provider, you will enjoy value-added features such as message waiting indication (MWI), hotline, three-way conferencing, caller ID, call forward (always, busy and no answer), call waiting, Do Not Disturb (DND), dial plan and phone book. Using a VIP-156PE with an analog phone as an IP

PBX telephone system gives you a special feature at a consumer price; the VIP-156/VIP-156PE/VIP-157S can be seamlessly integrated into the telephony network built by our IP PBX system series. The

VIP-156/VIP-156PE/VIP-157S and PLANET IP PBX System (IPX-330/IPX-2100/IPX-2200/IPX-2500) are the ideal combination of your office daily communications.

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Compliant with IETFSIP 2.0


SIP service continues to gain popularity among businesses as the preferred protocol for enhancing communication across IP networks. The VIP-156/VIP-156PE/VIP-157S supports Session Initiation Protocol

2.0 (RFC 3261), STUN and Outbound Proxy for easy integration with general voice over IP system. The

VIP-156/VIP-156PE/VIP-157S is able to broadly interoperate with equipment provided by VoIP infrastructure providers, thus enabling them to provide their customers with better voice over IP services.

Convenient Design Brings Quality Communication

Based on years of VoIP manufacturing experiences, PLANET VoIP total solutions are known as advanced implementation of standard-based telephony along with mass deployment capability.

The VIP-156/VIP-156PE/VIP-157S is the lowest cost, fastest deployment ready ATA with auto-provision. The

ATA can be deployed directly to customers, redirecting automatically to your provisioning server or you can use the TR069 ITSP portal for target profiles.

With the VIP-156/VIP-156PE/VIP-157S, enterprises are able to save the installation cost and extend their past investments in telephones, conference and speakerphones. The VIP-156/VIP-156PE/VIP-157S can be the bridge between the traditional analog systems and IP network with an extremely affordable investment.

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1.2 Application

Enhanced, Full-Featured Analog Telephone Adapter

The VIP-156/VIP-156PE/VIP-157S is optimized for executive use for administrative assistants and those working with bandwidth-intensive application on collocated PCs. Four programmable extension keys could be configured as IP PBX features like, MWI, DND, Call Forward, Call Park and many others.

IVR Function to Easily Identify and Manage the ATA

Through the Interactive voice response (IVR) function, user can simply press some function keys to search the device information or program the phone feature, e.g., #120 to check the LAN IP address, and #112 + xxx*xxx*xxx*xxx# to assign the LAN IP address.

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Enterprise IP Telephony Deployment of VIP-156PE


The VIP-156PE comes with exceptional audio quality and user-friendly features. The installation and configuration of the IP phone are easier than those of the traditional phone system. Its low cost and high-definition voice quality give you value for money. Based on standard SIP 2.0, it is compatible with all the standard SIP-based servers.

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1.3 Features



Product features

 Feature-rich telephone service over home or office Internet/Intranet connection

 Auto-provisioning features for flexible, ease-of-use IP PBX system integration

 IEEE 802.3af/at Power over Ethernet compliant (VIP-156PE)

 Up to 2 concurrent VoIP calls (VIP-157S)

 Voice prompt for machine configurations

 Quick Setup Wizard for easy configuration

 Easy access with PLANET Dynamic DNS

 Cost effective, field proven compatibility and stability



VoIP Feature

 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

 Voice processing: Voice Active Detection, DTMF detection/generation, G.168 echo cancellation

(16mSec.), comfort noise generation,

 In band and out-of-band DTMF support

 Auto-provision (FTPP, HTTP, FTP)



Other Features

 IPv6

 VPN connection

 Planet DDNS

 VLAN

 Local Phone book (download/upload)

 QoS

 IVR Function

 802.3af PoE integration (VIP-156PE)

 FXO integration (VIP-157)

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1.4 Package Contents

The contents of your product should contain the following items:

 VoIP Telephone Adapter

 Power Adapter

 Quick Installation Guide

 RJ11 Cable x 1

1.5 Physical Details

The following figure illustrates the front and rear panels of ATAs.

Respective models/descriptions are shown below:

 VIP-156: SIP Analog Telephone Adapter


Front Panel of VIP-156

Left and Right Panels of VIP-156

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 VIP-156PE: 802.3af PoE SIP Analog Telephone Adapter


Front Panel of VIP-156PE

Left and Right Panels of VIP-156PE

 VIP-157S: 2-port FXS SIP Analog Telephone Adapter

Front Panel of VIP-157S

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Left and Right Panels of VIP-157S

LED Display & Button

1

2

PC

LAN

RJ45 connector, to maintain the existing network structure, is connected directly to the PC through straight Cat5 cable.

RJ45 connector, for Internet access, connected directly to Switch/Hub through straight Cat5 cable.

The LAN interface also can be connected with 802.3af PoE switch or converter for power supply ( VIP-156PE ).

3

4

12V DC

Reset

12V DC Power input outlet

Reset to the factory default setting

Machine default IP is http://192.168.0.1. Press the RESET button on the rear panel for over 5 seconds to reset the VoIP Phone Adapter to factory default value. (Speed dial and call forward settings are exceptional.)

LED Display of VIP-156/VIP-156PE

LED Indicators Descriptions

PWR Power is supplied to the device.

STATUS

LNK/ACT

The Status LED will flash when the machine is operational.

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

ON : The device is connected to LAN at 100Mb/s.

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RING

OFF : The phone is idle.

ON : The phone is in use (off hook).

Blinking : The phone is ringing.

LED Display of VIP-157/VIP-157S


STATUS


The Status LED will flash when the machine is operational.

LNK/ACT

RING 1

RING 2

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

ON : The device is connected to LAN at 100Mb/s.







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CHAPTER 2. PREPARATIONS & INSTALLATION

2.1 Physical Installation Requirements

This chapter illustrates basic installation of ATA Analog Phone Adapter (“ ATA ”).

•

Network cables. Use standard 10/100BASE-TX 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 log on to the ATA web configuration page.

ATA will prompt for logon username/password. Please enter: root / null (no password) to continue machine administration.

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 the related chapter on user’s manual CD or consult your network administrator for proper network configurations.

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2.2 LAN IP address configuration via web configuration interface

Execute your web browser, and insert the IP address ( default: 192.168.0.1

) of VIP in the address bar. After logging on to the machine with username and password (default : root and no password ), browse “ Network ”

--> “ Network Settings ” configuration menu:

Parameter Description

IP Address

Subnet Mask

LAN IP address of ATA

Default : 192.168.0.1

LAN mask of ATA

Default : 255.255.255.0



Hint

Default Gateway 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.

When you want to run the setup or the start function, you must unlock the protect function #190# before setting up network settings and ATA function via keypad.

IVR Menu Choice Machine Operation

#111# Set DHCP client

Parameter(s) Notes:

None

ATA will change to

DHCP Client

Use the * (star) key DHCP will be disabled

#112xxx*xxx*xxx* xxx#

Setup Static IP Address decimal point. to the Static IP type.

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Parameter(s)


Notes:

Use the * (star) key

Set Network Mask decimal point.


Set Gateway IP

Address decimal point.


#190#

#191#


Set Primary DNS when entering a

Server decimal point.

Unlock None

Must unlock the protect function before setting up network settings and

ATA function via keypad.

Lock None

The system will be locked and can’t set up network settings via keypad.

#195#

#198#

Reboot

Factory Reset

None

None

The system will reboot automatically.

The system will be reset to factory default value and reboot automatically.

0 ＊

To switch PSTN mode None VIP-157 only

The following keypad commands can be used to display the network settings enabled on ATA via voice prompt.


#120# Check PC IP Address

Notes:

IVR will announce the current PC-port

IP address of the ATA.

#121#

#122#

#123#

#124#

Check network connection type

Check the Phone Number

Check Network Mask

Check Gateway IP Address

IVR will announce if DHCP is enabled or disabled.

IVR will announce current enabled VoIP number.

IVR will announce the current network mask of the ATA.

IVR will announce the current gateway


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IVR Menu Choice

#125#

#126#

Machine Operation


Notes:

Check Primary DNS Server IVR will announce the current setting in

Setting the Primary DNS field.

IVR will announce the current LAN port

Check LAN IP Address


#128#

IVR Menu Choice

Check Firmware Version

Machine Operation

IVR will announce the version of the firmware running on the ATA.

The following keypad commands can be used to set up the main function .

Parameter(s) Notes:

#138#

#139#

#160#

#510#

#511#

#512#

#514#

#130+[1~8]#

#131+[00~15]#

#132+[00~12]#



#137+[0~2]#

Enable call waiting

Disable call waiting

Update firmware

Blind Transfer

Attendant Transfer

3-way calling

IP transfer to PSTN

Set Codec

Set Handset Gain

Set Handset Volume

TFTP Server IP Address

FTP Server IP Address

Auto config mode

None

None

None

ATA Blind Transfer

ATA Attendant Transfer

Enable call waiting

Disable call waiting

Update firmware

ATA 3-way calling

ATA transfer IP call to

PSTN side

1:G.711 u-Law, 2:

G.711 a-Law, 3:

G.723.1, 4: G.729a, 5: You can set the codec you

G.726 16K, 6: G.726 want to be the first priority.

24K, 7: G.726 32K, 8:

G.726 40K,

Handset Gain from

You can set the Handset gain to proper value,

0~15 default is 10

Handset Volume from

You can set the handset volume to proper value;

0~12 default is 10

Set Auto config TFTP You can set the TFTP

Server IP Address Server IP address

Set Auto config FTP You can set the FTP

Server IP Address Server IP address

You can set the auto

0: Disable, 1: TFTP

Disable, 1: use TFTP mode, 2: FTP mode configuration mode, 0:

Server, 2: user FTP

Server

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IVR Menu Choice

#145#


Machine Operation Parameter(s)

Forward function disable Disable forward function

Notes:

#146+Number#

Enable forward to FXS Eanble forward to FXS

Port Port

#147+Number#

#116#

#117#

#118#

#119#

Enable forward to FXO Eanble forward to FXO

Port Port

Enable PPTP function None

Disable PPTP function None

Enable VLAN function None

Disable VLAN function None

Enable PPTP function

Disable PPTP function

Enable VLAN function

Disable VLAN function



Hint

Please contact your Internet service provider to obtain the Internet access type, and select the proper network settings in ATA to establish the network connections.

After confirming the modification you’ve done, please click on the Submit button to apply settings and browse

“ Save & Reboot ” menu to reboot the machine to make the settings effective.

Connection Type Data Required.

Fixed IP

DHCP client

PPPoE

In most circumstances, it is no need to configure the DHCP settings.

The ISP will assign IP Address, and related information.

The ISP will assign PPPoE username/password for Internet access,



Hint

Please consult your ISP personnel to obtain proper PPPoE/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.

Save Modification to Flash Memory

Most of the VoIP router parameters will be effective after modifications, but it is just temporarily stored on

RAM only. It will disappear after you reboot or power off the VoIP Phone Adapter. To save the parameters into

Flash ROM and let it be effective forever, please remember to press the Save & Reboot button after you modify the parameters.

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CHAPTER 3. NETWORK SERVICE

CONFIGURATIONS


3.1 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.

3.1.1 Overview of the web interface of ATA

With web graphical user interface, you may have:

 More comprehensive setting feels than traditional command line interface.

 Input data fields, check boxes, and changing machine configuration settings

 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

3.1.2 Manipulation of ATA via web browser

Log on to 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 log on to the Phone Adapter web configuration page.

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

ATA login page

When users log in to the web page, users can see the Phone Adapter system information like firmware version, company, etc on this main page.

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VoIP Phone Adapter main page

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CHAPTER 4. VOIP TELEPHONE ADAPTER

CONFIGURATIONS

4.1 Status

Status shows all the system information like WAN/LAN IP address, system information, IPv6 connection information, register status and VPN connection message. (After you set up the VPN line, the status will start to show.)

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4.2 Phone Book

ATA can set up 140 records of Phone Book. User can dial the Name records to make calls via Phone Book feature.

Field

Phone Book Page

Phone

Description

The default is Page 1. You can select Page 1 ~ Page 7 to look through

Phone Book records.

The record number is from 1 ~ 140; it can set up 140 records in total.

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Description

The name of Phone Book records; it only can input numerals character.

Fill in the outgoing number (Line Number) or IP address.

Clean this item’s data.

Save the phone book data as CSV file.

Upload the phone book file


Field

Name

URL

Delete

Export csv

Upload

If you need to add a phone number to the Phone Book list, you need to input the position, the name, and the phone number (by URL type). When you finish a new phone list, just click the “Submit” button.

If you want to delete a phone number, you can select the phone number you want to delete and then click the

“Delete” button.

Press “Reset” to erase the data that you didn’t save.

For Example:

Example_1:

ATA has added the above phone numbers. User picks up the handset and dial “ 301 ” to make the P2P call

( [email protected]

).

Example_2:

User picks up the handset and dial “ 206 ” to make the Proxy call (17476433364).

Example_3:

User picks up the handset and dial “ 202 ” to make the P2P call (192.168.1.2:5062).

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4.3 Call Service


[Call Forward]

This page defines Call Forward function. You can set up the phone number you want to forward on this page.

There are three types of Forward mode. You can choose All Forward, Busy Forward, and No Answer Forward by clicking the icon.

All Forward :

All incoming call will forward to the number you choose. You can input the name and the phone number in the

URL field. If you select this function then all the incoming calls will direct forward to the speed dial number you choose.

Busy Forward :

If you are on the phone the new incoming call will forward to the number you choose. You can input the name and the phone number in the URL field.

No Answer Forward :

If you cannot answer the phone, the incoming call will forward to the number you choose. You can input the name and the phone number in the URL field. Also you have to set the Time Out time for system to start to forward the call to the number you choose.

When you finish the setting, please click the Submit button.

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[Hotline Type]


This page defines the Hot line setting on this page. When user picks up the handset, the device will call to the specific number automatically.

Hotline Type:

Click Enable to carry the Hot line function out.

Hotline number:

The hot line number can input the IP address or registration number.

Delay time:

If you don’t dial for a period of time, it will automatically dial the hotline number.

[DND Type]

This page defines the DND Setting to keep the phone silent. You can choose Always Block or Block a period.

Always Block :

All incoming calls will be blocked until this feature is disabled.

Block Period :

Set a time period and the phone will be blocked during the time period. If the “From” time is larger than the

“To” time, the Block time will from Day 1 to Day 2.


[Alarm Type]

This page defines the Alarm setting on this page. It provides the alarm function, and it can set up the Alarm

Time to get the telephone rung every day.

Alarm Type:

The default is Off. If set up as On, the telephone will ring at a specific time.

Alarm Time:

It can set up the system prompt time within 24 hours.

Alarm Line:

Select the Line for alarm.(only for VIP-157S)

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4.4 SNTP setting

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, depending on how long you need to synchronize it again.

User can also use the “daylight saving” to adjust the daylight time. When you finish the setting, please click the Submit button.

4.5 Volume Setting

This page defines the Handset Volume, Ringer Volume, and the Handset Gain. When you finish the setting, please click the Submit button.

Handset Volume is to set the volume you can hear from the handset.

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

Volume Settings for VIP-156T/VIP-156PT

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4.6 Dial Plan Setting

This page defines the Dial Plan Setting function. This function is to input the phone number by the keypad but you don’t need to press “#”. After time out the system will dial directly.

Field

Drop Prefix

Prefix

Rule Rule

Dial Plan Settings for VIP-156

Description

The rule of add or replace code -- If Disable is set, it will add the prefix number prior to the identification number. If Enable is set, it will replace the identification number.

The prefix number -- It only accepts 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 accepts the [+], [x]

- +: It means “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 begins with 5.

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Field


Description

If the dialing number matches this field, it will dial out and need not have to press the “#” key to end the dialing. It accepts the numeral or symbol, and the max. length are 124.

Dial Now Rule

The starting number can’t be “0”. For example, if the number is “0xxxx”, the system will ignore this dial plan.

Realm 1/2/3/4/5

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

Area Code

Inter Digit Time

Dialed number is sent out after dialing is stopped.

(Auto Dial Time)

If Yes is set, the system will stop receiving the dialed number when receiving the [#] key. The system also will determine the Auto Dial Time,

Key as send # It will carry out the calling if there is no dialing after the Auto Dial Time. If

No is set, the system will do according to the Auto Dial Time to determine the end time.

Descriptions of example:

Example_1: Drop prefix: Disable , Prefix: 002 , Rule: 8613+8662

1. If the dialing number is “ 8613xxxxx ”, it will match the rule [8613], then system will automatically add the prefix [002] in front of [8613].The real dialing number is [ 002+8613xxxxx ].

2. If the dialing number is “ 8662xxxxx ”, it will match the rule [8662], then system will automatically add the prefix [002] in front of [8662].The real dialing number is [ 002+8662xxxxx ].

Example_2: Drop prefix: Enable , Prefix: 006 , Rule: 002+003+004+005+007+009

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1. If the dialing number is “ 002+86xxxx ”, it will match the rule [002], then system will automatically replace the prefix [002] to the prefix number [006].The real dialing number is [ 006+8613xxxxx ].

2. If the dialing number is “003+77 xxxx ”, it will match the rule [003], then system will automatically replace the prefix [003] to the prefix number [006]. The real dialing number is [ 006+77xxxx ].

Example_3: Drop prefix: Disable , Prefix: 009 , Rule: 12

1. If the dialing number is “ 12xxxxx ”, it will match the rule [12], then system will automatically add the prefix

[009] in front of [12].The real dialing number is [ 009+12xxxxx ].

Example_4: Drop prefix: Disable , Prefix: 009 , Rule: 53+35xx+21xx

1. If the dialing number prefix is [53789], it will match the rule [53], then system will automatically add the prefix [007] in front of [53789].The real dialing number is [ 007+53789 ].

2. If the dialing number prefix is [3507], it will match the rule [35xx], then system will automatically add the prefix [007] in front of [3507].The real dialing number is [ 007+3507 ].

3. If the dialing number prefix is [2199], it will match the rule [21xx], then system will automatically add the prefix [007] in front of [2199].The real dialing number is [ 007+2199 ].

Example_5: Dial Now: *xx+#xx+11x+xxxxxx

1. If the dialing number matches with the rule of “ *xx ”, it will send out the dialing number directly. For example, *00/ *01/ *02…*99 .

2. If the dialing number matches with the rule of “ #xx ”, it will send out the dialing number directly. For example, #00/ #01/ #02…#99 .

3. If the dialing number matches with the rule of “ 11x ”, it will send out the dialing number directly. For example, 111/ 112/ 113…119 .

4. If the dialing number matches with the rule of 8 digits, it will send out the dialing number directly. For example, 12345678 .

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4.7 General

This page defines the volume, auto answer, caller ID, and call waiting caller ID (CID type II),

VIP-156

Field

Call Waiting

Ring Timeout

Caller ID Scheme

Description

Default is enable.

When you are talking with other people, you can choose if you want to hear the message when there is a new incoming call. If the call waiting function is On, if there is a new incoming call, you will hear the call waiting message in your current call. If you set the function to Off, then you will not hear any message.

Default is 60 (sec). After how long the system will reply the busy (486 busy) message.

Set the caller ID mode, it supports FSK Bellcore, DTFM, CID-Japan,

DTMF-Brazil, DTMF-Denmark.

FSK Bellcore: FSK caller ID mode.

DTMF: Before the first ring, it will send the DTMF caller ID data.

CID-Japan: Japan (Japan) caller ID mode

DTMF-Brazil: Brazil (Brazil) caller ID mode

DTMF-Denmark: Denmark (Denmark) caller ID mode

To enable the show caller ID function in call waiting.

When enabling this function, system will receive a new call in call waiting. It will display the caller ID

CID Type II

T.38 (Fax)

T.38 Pass-through

Codec a-law.

Your phone must also support CID Type 2.

E nable/Disable T.38 Fax function.

Define the T.38 pass-through codec; it can support G.711 u-law/G.711

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CHAPTER 5. NETWORK

5.1 Network Setting

This page defines the LAN setting on this page.


Field

WAN Active

Description

The default is Fixed IP, and it also provides DHCP Client and PPPoE connection modes.

Fixed IP: It could set up the IP address manually.

DHCP Client: It will acquire the IP address automatically.

PPPoE: It will use the PPPoE connection method

The IP address IP Address

Subnet Mask

Default Gateway

DNS Active

Primary DNS

Second DNS

MAC Address

System Name

The sub net address

The default gateway address

Static/Automatically, manually set up the DNS server or automatically accept the DNS server.

The default is 168.95.192.1; it could set up the first DNS server address.

The default is 168.95.1.1; it could set up the second DNS server address.

The MAC of LAN port

PPPoE User Name

PPPoE Password

The product model

The PPPoE connection account name. It could input numeral or character; the maximum date length is 63.

The PPPoE connection account password. It could input numeral or character; the maximum date length is 63.

PPPoE Service name PPPoE Service provider name

PPPoE AC Name PPPoE AC name.

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5.2 DDNS Setting


This page defines the DDNS setting on this page. You need to have the DDNS account and input the information properly. You can have a DDNS account with a public IP address, then others can call you via the

DDNS account. But now most of the VoIP applications work with an SIP Proxy Server. When you finish the setting, please click the Submit button.

(For better service, Planet provides Planet DDNS . You can apply your DDNS account on its web site www.planetddns.com)

DDNS Settings for VIP-156/VIP-156PE/VIP-157S

5.3 VLAN Setting

This page defines the VLAN setting on this page. This function needs to co-operate with network devices which have VLAN function. Also this page defines the SIP and RTP port number on this page. Each ISP provider will have a different SIP/RTP port setting. Please refer to the ISP to set up the port number correctly.


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Field

VLAN Activity

VID (802.1Q/TAG)

User Priority (802.1P)

Description

If On is set, it could receive VLAN messages.

Dispose VLAN ID is added as a Tag header after enabling the

VLAN function. The realized voice packets transfer is similar to that of VLAN. The prerequisite is it must be the same as VLAN of upper switch. The value range is 2~4094.

To set up the user priority.

Field

SIP VID

SIP User Priority (802.1P)

RTP VID

RTP User Priority (802.1P)

Description

Set the SIP VLAN ID; this is an independent function that doesn’t need to enable [VLAN Packets: Enable].

Set up the SIP Priority.

Set the SIP RTP VID; this is an independent function that doesn’t need to enable [VLAN Packets: Enable].

Set up the RTP Priority.

5.4 VPN Setting

This page defines the PPTP/L2TP setting on this page. You could set up the PPTP/L2TP Server connection information. When you finish the setting, please click the Submit button.

Caution: VIP-156/VIP-157 series VPN can’t use the encryption or compression for VPN connection.

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5.5 IPv6 Setting


This page defines the IPv6 setting on this page. You can program the IPv6 information.

Field

IPv6 Activity

IPv6 address

SubnetPrefix Length

Default Gateway

LAN IPv6 Address:

Description

Support three IPv6 types: Auto, Fixed IPv6, IPv6 in IPv4 Tunnel

Setting the WAN IPv6 address or display it (64 bits)

Default is 64

IPv6 gateway address (64 bits)

IPv6 LAN address (64 bits)

LAN IPv6 Link-Local Address Link local address information.

Autoconfiguration Type It supports stateless or stateful (DHCP V6).

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CHAPTER 6. NAT TRANS

6.1 Stun Setting

This page defines the STUN Enable/Disable and STUN Server IP address on this page. This function can help your Phone Adapter work properly behind NAT. To change this setting, please follow your ISP information. When you finish the setting, please click the Submit button.

6.2 PC Setting

This page defines the PC setting on this page.

Field

Device Activity

PC IP address

Description

The default is Bridge mode, and it also provides NAT mode.

Bridge: When set as mode, the LAN and PC ports are in the same network segment.

NAT: The LAN and PC ports are in a different network segment, and PC port could enable the DHCP Server function to allot the IP address.

The IP address of PC port. (In the Birdge mode, the Default IP :

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Field Description

PC MAC Address

192.168.0.1

The MAC of PC port

Enable DHCP Server It will allot the IP address automatically when enable this function.

IP Address

Lease Time

The range for DHCP IP address.

DHCP server lease time

6.3 DMZ and MAC Clone

This page defines the DMZ and MAC Clone setting on this page.

DMZ Activity: If set up as On, all of the packets (except SIP packets) will be sent to the specific IP address.

DMZ IP Address: The DMZ host IP address.

MAC Clone Activity: This page defines the MAC Clone Enable/Disable. This function will copy the MAC address from NIC (Network Interface Card) which is placed in PC to LAN port of ATA. That is because some

ISPs will limit the MAC address for PPPoE dial-up connection.

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6.4 Virtual Server

This page defines the Virtual Server setting on this page. You could define 24 virtual service information on this page. When you finish the setting, please click the Submit button.

Field

Index

Activity

Protocol

Internal Port

External Port

Server IP

Delete

Description

The serial number -- There are totally 12 records from Num 1 to 12.

The activity status -- The default is Disable; this record will activate if enable.

The TCP or UDP communication protocol.

For corresponding the internal port.

For corresponding the external port.

To input the Server IP address.

Delete this item

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CHAPTER 7. SIP SETTING

7.1 Service Domain Setting

In Service Domain function, you need to input the account and the related information on this page. Please refer to your ISP provider. You can register five SIP accounts in the ATA. You can dial the VoIP phone to your friends via first enable SIP account and receive the phone from these five SIP accounts.

Field

Realm

Realm Activity

Description

Which line you want to use.

First you need to click Active to enable the Service Domain, then you can input the following items.

Display Name

Phone number

The serial number -- There are totally 24 records from Num 0 to 23.

The activate status -- The default is Disable; this record will activate if enabled.

You need to input the Register Password obtained from your ISP. Authentication ID

Authentication Password You need to input the Register Name obtained from your ISP.

Domain Server You need to input the Domain Server from your ISP.

Proxy Server You need to input the Proxy Server from your ISP.

Outbound Proxy

You need to input the Outbound Proxy from your ISP. If your ISP does not provide the information, then you can skip this item.

When system is enabled, it will frequently send the MWI message.

Subscribe to MWI

Note

The starting number can’t be “0”. For example, if the number is

“0xxxx”, the system will ignore this dial plan.

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You can see the Register Status on the Status page. If the item shows “Registered”, then your Phone Adapter is registered to the ISP, and you can make a phone call directly.

If you have more than one SIP account, you can follow the steps below to register to the other ISP.


7.2 Codec Setting

This page defines the Codec priority, RTP packet length, and VAD function on this page. You need to follow the ISP suggestion to set up these items. When you finish the setting, please click the Submit button. Also on this page, it defines the Codec ID. Sometimes two VoIP devices with different Codec IDs will cause the interoperability issue. If you are talking with some people with some problems, you may ask the other one what kind of Codec ID he uses, and then you can change your Codec ID. When you finish the setting, please click the Submit button.

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7.3 SIP Advance Setting

This page defines the Hold by RFC, Voice/SIP QoS and other settings on this page. To change these settings, please follow your ISP information. When you finish the setting, please click the Submit button.

Field

SIP Expire Time

Type

Description

To set up the registration interval time.

General (standard), 1/2, 2/3, 3/4, 4/5, 5/6, 6/7, 7/8, 8/9, 9/10.

Note

Registered server needs to support this function.

Register time calculated

General: expiry time-[(expiry time/30)*6], when Expiry Time > 60 it will start to work; if less than 60 seconds, it will decrease 5 seconds.

1/2: expiry time * 1/2.

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SIP Register Retry

If SIP register fails, system will retry interval after this time.

Timer

SIP session timer T1 Setting the maximum retransmit interval for non-INVITE requests and

INVITE responses.

Note


SIP session timer T2 Setting the maximum retransmit interval for non-INVITE requests and

INVITE responses.

Note


SIP session timer Setting the maximum retransmit interval for non-INVITE requests and

Timer B, F, H INVITE responses.

Note


B: 64 * SIP T1; INVITE transaction timeout timer.

F: 64 * SIP T1; non-INVITE transaction timeout timer.

H: 64 * SIP T1, Wait time for ACK receipt.

Local SIP Port of Setting the phone 1 SIP start and end port. All the port can’t be phone 1 duplicated

Local RTP Port of Setting the phone 1 RTP start and end port. All the port can’t be phone 1 duplicated

Hold type The default is disable, and to start up communication hold back function (RFC definition). Set enable to start up the Hold by RFC function.

DTMF Mode Defines the InBand, RFC2833, SIP Info, RFC2833 + Inband, SIP Info

+ Inband on this page. To change this setting, please follow your ISP information. When you finish the setting, please click the Submit button.

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Field

RPort


Description

To change this setting, please follow your ISP information. When you finish the setting, please click the Submit button.

Note


Voice QoS

The Voice QoS feature.

(Diff-Serv)

SIP QoS (Diff-Serv) The SIP QoS feature.

The QoS setting is to set the voice packets’ priority. If you set the value higher than 0, then the voice packets will get the higher priority to the

Internet. But the QoS function still needs to cooperate with the other

Internet devices.

RTP Traffic Class

IPv6 RTP traffic class

(IPv6)

SIP Traffic Class

IPv6 SIP traffic class

(IPv6)

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

Always transport the network packets to keep the NAT port

Packet

Keep Alive Period

Jitter Buffer

To set up the interval time for transporting packets.

SIP Server Type

To set up the size for jitter buffer packets.

Provides a different register server: General, Asterisk, BroadWorks,

Nortel, Xener, Vodtel, SKTelink, for different server systems will adjust some system parameters

Note


Use user = phone When sending the registered package, in package Header will add

(Register): the ”user=phone” message.

Note


Use user = phone When sending the dialing package, in package Header will add

(Invite): the ”user=phone” message.

Note


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Send SIP PRACK to When sending the SIP package, in package Header will add

Proxy: the ”PRACK” message.

Note


Only Accept Trusted Only accept call from proxy; if system receives the IP dialing, system

Certificates: will refuse the call.

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CHAPTER 8. ADVANCE SETTING


8.1 Status Log

Display and save systems running status message data. Press “Get Status Log” to back up the status log file.

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8.2 Auto Config


This page defines the Auto Configuration (Auto Provision) setting. ATA supports TFTP, FTP, HTTP and IP

PBX auto configuration function in total. In IP PBX Auto Configuration Setting, you need to check with your service provider if they have provided this function.

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8.3 Management -- Advanced Setting


This page defines the advanced functions. When you finish the setting, please click the Submit button.


ICMP Not Echo

This function can disable echo when someone pings this device. It can avoid hacker trying to attack the device

If this function is enabled, machine will to start the calling hidden function, and it will not send the related Caller information.

Anonymous Call

Note


Management form

When [Enable] allow user login from WAN.

WAN

Stop Feature Tone

When [Enable] if system sets the function like [Subscribe to MWI, forward, DND], when user picks up the phone, he will hear the remind tone [Do Do Do]

There are three billing types: Polarity Reversal, Tone_12K and

Tone_16K.

Billing Signal

Note


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Field

CPC Delay

CPC Duration

IP Dialing Format

Send Flash event


Description

When receiving the disconnected signal, machine will cut the voltage down to 0V after this time.

When starting to cut the voltage down to 0V, machine will continue this state by this time.

Setting IP dialing format; when [Disable] can’t use IP dialing to make a call.

There are two flash formats: DTMF Event and SIP Info.

There are seven encrypt formats: Disable, INFINET, AVS,

WALKERSUN1, WALKERSUN2, CSF1, CSF2, GX, VGX, RC4,

VOS_R, VGCP.

Encrypt Type

Note


Some encrypt types must enter the Encrypt Key

Encrypt Key

Note


PPPoE Retry Period

If PPPoE dial-up connection fails, machine will retry the dial-up motion after this time.

DHCP Gateway ARP

The period to check the DHCP gateway ARP.

Check Period

Syslog Server IP

There are seven Syslog types: Call Statistics, General Debug, Call

Address

Statistics + General Debug, SIP Debug, Call Statistics + SIP Debug,

General Debug + SIP Debug and All.

System Log

Machine could send the system logs to the specific Syslog Server. It can input the IP or Domain address

PSTN port Country Set up the FXS Port Coutry

Timeout

PSTN CID forward disconnect the line.

It must work with [Phone – General] [Auto Answer] function or

[Phone – Caller Service] [Forward] function. When this function is enabled, the caller ID from FXO can transfer to other devices.

Generate Flash

FXO flash time defines to hold or hang up the phone

Signal for PSTN

FXS Port Coutry Select the FXO port local country

Flash Hook Time

Maximum flash time to detect the call on hold or hang up.

(Max)

Flash Hook Time

Minimum flash time to detect the call on hold or hang up.

(Min)

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NET Bandwidth

Setting the limitation for LAN bandwidth

Limit


8.4 Tones

This page defines the Tone settings. This function can set up the related parameters of Dial Tone, Ring Back

Tone, Busy Tone, Error Tone and other Tones. When you finish the setting, please click the Submit button.

8.5 TR-069

On this page you can program the TR-069 setting.

A different TR-069 server may need to modify some different parameters.

What’s TR-069: Technical Report 069 (TR-069) is a customer-premises equipment WAN

Note management protocol (CWMP) technical specification for remote management of end-user devices introduced by the broadband forum (formerly the DSL forum).TR-069 is an integrated framework equipped with safe auto-configuration. It also can take control of other CPE functions.

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CHAPTER 9. OTHER SETTINGS


9.1 System Authority

In System Authority, admin/System/User login password can be changed.

9.2 Firmware Upgrade

This page defines the SIP and RTP port number on this page. Each ISP provider will have different

SIP/RTPport settings. Please refer to the ISP to set up the port number correctly. When you finish the setting, please click the Submit button.

If your updated file is xxxx.ROM, you must enter http://VIP-15X’s-IP Address/update.htm

e.g. http://192.168.0.157/update.htm. to upload the ROM file and then update the system.

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For technological consideration, we strongly suggest you refer to the following upgrade methods for updating your device.

Note

After firmware is loaded, the unit will reboot, and default IP address of the customized firmware: http://192.168.0.1

; login name/password: root/null (no password)

9.3 Auto Update Setting

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

Field Descriptions

Type

TFTP Server

TFTP File Path

HTTP Server

HTTP File Path

There are TFTP/FTP and HTTP 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.


FTP Server

FTP Username

FTP Password

FTP File Path

Check new firmware

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

Name form.

The login username.

The login password


The device will do according to the ways below to check the new firmware.

– Power On (+ Scheduling) : The machine will check the new firmware when power on and following scheduled date and time.

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Field

Scheduling (Date)

Automatic Updat e


Descriptions

– Scheduling : The machine will follow the scheduled date and time to check the new firmware.

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

“Beep” sound when picking up the handset to prompt you there is a new firmware.

– Automatic : The device will carry out firmware update automatically.

Firmware File Prefix It will check the information on model name.

Next update time It will show the next check date and time.

If the Check new firmware field is set to Power On, the machine will check the new firmware according the scheduled time/date and power on. If there is a new firmware, it can be

Note upgraded. The machine won’t carry out firmware update automatically The machine will send the prompt sounds when picking up the handset, and it needs to update firmware manually.

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9.4 Reset to default


In Default Setting, you can restore the Phone Adapter to factory default on this page. You can just click the

Restore button, then the Phone Adapter will restore to default and automatically restart again.

9.5 Save and Reboot

In Save & Reboot, you can save the changes you have done. If you want to use new setting in the Phone

Adapter, you have to click the Save button. After you click the Save button, the Phone Adapter will automatically restart and the new setting will take effect.

9.6 Logout

Logging out the system will return you to the login page.

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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 addresses are 192.168.0.1 and 192.168.0.2

Analog telephone sets are connected to the phone (RJ11) 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# and will 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

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 ’, and phone 2 will ring.

Analog telephone sets are connected to the phone (RJ11) ports of VIP-157S respectively

192.168.0.1

1001 1002

1 9 2 * 1 6 8 * 0 * 1 * * 5 0 6 2 #

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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# and will 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.

 If the IP address of the remote calling party is known, you may directly make calls via its IP address and end with a “ # ”.

 If the ATAs are installed behind a NAT/firewall/IP sharing device for Peer-to-Peer VoIP application, please make sure the NAT device supports SIP applications, and suitable settings should be applied to the NAT device to enable the SIP communications before making calls

 [ 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

Case 3: Call Forward Feature_Example 1

In the following samples, we’ll introduce the Call Forward Feature applications.

For this example, there are three VIP-156 registers to IPX-1980 and VIP-156_A has set Call Forward function to VIP-156_B.

Machine configuration on the VIP-156:

Please log in to VIP-156_A via web browser and browse the Phone Settings menu and select the Call service config menu. On the setting page, please enable the All Forward function and fill in the Forward

Type and Forward Number of VIP-156_B, then the sample configuration screen is shown below:

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1. VIP-156_C picks up the telephone

2. Dial the number 1001 (VIP-156_A),

3. Because VIP-156_A has set up All Forward function to the number 2002 (VIP-156_B)

4. The number 2002 (VIP-156_B) will ring up; then it picks up the telephone and communication with the number 3003 (VIP-156_C).

Case 4: Call Forward Feature_Example 2

For this example, there are three VIP-156 and connect with Peer to Peer mode. VIP-156_A has set Call

Forward function to VIP-156_B.

Machine configuration on the VIP-156:

Please log in to VIP-156_A via web browser and browse the Phone Settings menu and select the Call service config menu. On the setting page, please enable the All Forward function and fill in the Forward

Type and Forward Number of VIP-156_B, and then the sample configuration screen is shown below:


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 has set up All Forward function to the IP Address 192.168.0.2 (VIP-156_B)

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4. The IP Address 192.168.0.2 (VIP-156_B) will ring up; then it picks up the telephone and communication with the VIP-156_C

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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 called A and they are in the process of conversation.

2. A carries out the transfer function (Press the “ transfer ” button) to hold the conversation with B.

3. A presses “ #510# ” and hears the dial tone and then input the number of C (Followed by the “ # ” key).

4. C will ring up and A will get the busy tone for prompting to hang up

5. C picks up the handset and has conversation with B.

B. Attendant Transfer

1. B called A and they are in the process of conversation.

2. A carries out the transfer function to hold the conversation with B.

3. A presses “ #511# ” and hears the dial tone and then input the number of C (Followed by the “ # ” key).

4. C will ring up.

5. C picks up the handset and has conversation with A.

6. A hangs up and C has conversation with B.

3-Way Conferencing

1. A and B are in the process of conversation.

2. A wants to invite C to join their conversation.

3. A presses “ Transfer ” or “ Hold ” button to hold the conversation with B first and then press “ #512# ” and hear the dial tone, and then input the number of C (plus the “ # ” key).

4. C will ring up and pick up the handset to have conversation with A.

5. A presses the “ Transfer ” button again, and they will enter the 3-way conference mode.

Call Waiting

1. A and B are in the process of conversation.

2. C called A and A will hear the prompt sounds.

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3. A presses the “ Hold ” button to hold the conversation with B, and switch to have 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 anyone’s SIP accounts for making calls through inputting the switch code.

Realm switch code:

1* : Realm 1

2* : Realm 2

3* : Realm 3

4* : Realm 4

5* : Realm 5

For example, the default is realm 1, input the 2* (Followed by the # key) from keypad and hang up the telephone set. It will switch to realm 2, and it can make the SIP calls via realm 2.

Auto Update firmware manually (Keypad)

If you pick up the handset of ATA, you will hear the “DoDoDo” prompt. If want to carry out the upgrade action, please input ” #190# ” to unlock the device first. Then input ” #160# ” to upgrade the new firmware.

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APPENDIX C VIP-156/VIP-156PE/VIP-157S

SPECIFICATIONS

Product

Model

Hardware

SIP Analog Telephone Adapter

VIP-156 VIP-156PE VIP-157S

LAN

PC

1 x 10/100Mbps RJ45 port (802.3af PoE for VIP-156PE)

1 x 10/100Mbps RJ45 port

FXS (for telephone set connection)

1 x RJ11

Protocols and Standard

2 x RJ11

Standard

Voice codec

Fax support

SIP 2.0 (RFC3261)

G.711 u-law/a-law, G.723 (6.3k/5.3k), G.726, G.729

T.38, G.711 pass-through

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), IPv4/IPv6 TCP/IP, UDP/RTP/RTCP, HTTP, ICMP, ARP,

DNS, DHCP, NTP/SNTP, PPP, PPPoE, VLAN, VPN (PPTP & L2TP), TR069

Network and Configuration

Access Mode

Management

Static IP, PPPoE, DHCP

Web, keypad

Dimensions (W x D x H) 94 x 72 x 30 mm

Operating Environment 0~40 degrees C, 10~95% humidity

Power Requirements 12V DC

EMC/EMI CE, FCC Class B

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APPENDIX D PLANET DDNS APPLICATION

Configuring PLANET DDNS steps:

Step 1 Enable DDNS option through accessing web page of ATA device.

Step 2 Select DDNS server provided, and register an account if you have not used yet.

Let’s take dyndns.org as an example. Register an account at http://planetddns.com

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