ATA User`s manual - PLANET Technology Corporation.

ATA User`s manual - PLANET Technology Corporation.
SIP Analog Telephone Adapter
VoIP Analog Telephone
Adapter
VIP-156 / VIP156PE / VIP-157 / VIP-157S
1
SIP Analog Telephone Adapter
Copyright
Copyright© 2013 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. 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.
Federal Communication Commission 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
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SIP Analog Telephone Adapter
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.
R&TTE Compliance Statement
This equipment complies with all the requirements of DIRECTIVE 1999/5/EC OF THE EUROPEAN
PARLIAMENT AND THE COUNCIL OF 9 March 1999 on radio equipment and telecommunication terminal
Equipment and the mutual recognition of their conformity (R&TTE) The R&TTE Directive repeals and
replaces in the directive 98/13/EEC (Telecommunications Terminal Equipment and Satellite Earth Station
Equipment) As of April 8, 2000.
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:
♦ 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 for PLANET VoIP Analog Telephone Adapter
Model: ATA model
Rev: 1.1 (June, 2013)
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SIP Analog Telephone Adapter
TABLE OF CONTENTS
Chapter 1 Introduction............................................................................................................ 6
Overview....................................................................................................................................................... 6
Package Contents......................................................................................................................................... 9
Physical Details ............................................................................................................................................ 9
LED Display & Button ...................................................................................................................... 12
Chapter 2 Preparations & Installation................................................................................. 14
Physical Installation Requirements.......................................................................................................... 14
LAN IP address configuration via web configuration interface......................................................... 15
Chapter 3 Network Service Configurations......................................................................... 20
Configuring and monitoring your ATA from web browser.................................................................... 20
Overview of the web interface of ATA............................................................................................... 20
Manipulation of ATA via web browser .............................................................................................. 20
Chapter 4 VoIP Telephone Adapter Configurations ........................................................... 22
Status.................................................................................................................................................. 22
Phone Book........................................................................................................................................ 23
Call Service........................................................................................................................................ 25
SNTP setting ...................................................................................................................................... 27
Volume Setting................................................................................................................................... 27
Dial Plan Setting ................................................................................................................................ 28
General............................................................................................................................................... 32
Chapter 5 Network................................................................................................................. 35
Network Setting ................................................................................................................................. 35
DDNS Setting .................................................................................................................................... 36
VLAN Setting .................................................................................................................................... 36
VPN Setting ....................................................................................................................................... 37
IPV6 Setting....................................................................................................................................... 38
Chapter 6 NAT Trans ........................................................................................................... 39
Stun Setting........................................................................................................................................ 39
PC Setting .......................................................................................................................................... 39
DMZ and MAC Clone ....................................................................................................................... 40
Virtual Server ..................................................................................................................................... 40
Chapter 7 SIP Setting ........................................................................................................... 42
Service Domain Setting ..................................................................................................................... 42
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SIP Analog Telephone Adapter
Codec Setting ..................................................................................................................................... 43
SIP Advance Setting........................................................................................................................... 44
Chapter 8 Advance Setting .................................................................................................. 49
Status Log .......................................................................................................................................... 49
Auto Config ....................................................................................................................................... 49
Management-Advanced Setting ......................................................................................................... 50
Tones.................................................................................................................................................. 52
TR-069 ............................................................................................................................................... 53
Chapter 9 Other Setting ...................................................................................................... 55
System Authority ............................................................................................................................... 55
Firmware Upgrade ............................................................................................................................. 55
Auto Update Setting........................................................................................................................... 56
Reset to default .................................................................................................................................. 58
Save and Reboot ................................................................................................................................ 58
Logout................................................................................................................................................ 58
Appendix A Voice Communication Samples ......................................................................................... 59
Case 1: ATA to ATA connection via IP address .................................................................................. 59
Case 2: (Peer-to-Peer mode) VIP-157S Port 1 to Port 2 communications ......................................... 60
Case 3: Call Forward Feature_Example 1.......................................................................................... 60
Case 4: Call Forward Feature_Example 2.......................................................................................... 61
Case 5: Call Forward Feature_Example 3.......................................................................................... 62
Case 6: Call Forward Feature_Example 4.......................................................................................... 63
Case 7: Auto Answer Feature_IP to PSTN......................................................................................... 64
Case 8: Auto Answer Feature_PSTN to IP......................................................................................... 66
Appendix B The method of operation guide............................................................................................ 68
Call Transfer ...................................................................................................................................... 68
3-Way Conference.............................................................................................................................. 68
Call Waiting ....................................................................................................................................... 68
Switch the Realm (Registration Proxy Server) .................................................................................. 69
Auto Update firmware manually (Keypad)........................................................................................ 69
Appendix C VIP-156/VIP-156PE/VIP-157/VIP-157S Specifications .................................................... 70
Appendix D Planet DDNS Application .................................................................................................... 71
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SIP Analog Telephone Adapter
Chapter 1
Introduction
1
Overview
Based on years of VoIP manufacturing experiences, PLANET Technology VoIP total solutions are
known for advanced implementation of standards based telephony with mass deployment capability.
Cost-effective, High-performance
Cost-effective, easy-to-install and simple-to-use, the 802.3af PoE integration(VIP-156PE) 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 device, home users and companies are able to save the installation cost and extend their
past investments in telephones, conference and speakerphones. The ATA device can be the bridge
between traditional analog systems and IP network with an extremely affordable investment.
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SIP Analog Telephone Adapter
Enhanced, Full-Featured VoIP Adapter
For the new generation communication age, the ATA device supports IPV6 and VPN connection to
provide users with more flexible and advantageous communication product. The ATA device and our IP
PBX system integration are the ideal combination for your office daily communications The VIP-156
supports all kinds of SIP based phone features including Phone book, Call forward (busy, no answer,
always), Dial plan setting, SNTP setting, DDNS, VLAN, VPN(PPTP&L2TP), IPV6, DMZ, Mac Clone,
virtual server…
Standard Compliance
Compliant with the Session Initiation Protocol 2.0 (RFC 3261), the ATA device is able to broadly
interoperate with equipment provided by VoIP infrastructure providers, thus enabling them to provide
their customers with better multi-media exchange services.
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SIP Analog Telephone Adapter
IVR Function to Easily Identify and Manage the ATA
Through the Interactive voice response (IVR) function, user can simply press some function key to
search the device information or program the phone feature, e.g #120 to check the LAN IP address,
#112 + xxx*xxx*xxx*xxx# to assign the LAN IP address…..
Through Auto Provision to Synchronize Configuration Parameters
Through TFTP, FTP or HTTP auto provision function, user can Synchronize Configuration Parameters of
the ATA device at the same time. User can easily control the parameters and configuration of thousands
of ATA device from the web interface, without having to provision each unit individually.
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SIP Analog Telephone Adapter
Features
¾ Product features
„
Feature-rich telephone service over home or office Internet/Intranet connection
„
Cost effectiveness, field proven compatibility, and stability
„
Web-based and telephone keypad machine configuration
„
Remote administrator authentication
„
Voice prompt for machine configurations
„
DMZ and MAC clone
¾ 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)
¾ 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)
Package Contents
„
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.
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SIP Analog Telephone Adapter
„
Respective models/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
Left / Right Panel of VIP-156
Front Panel of VIP-156PE
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SIP Analog Telephone Adapter
Left / Right Panel of VIP-156PE
Front Panel of VIP-157
Left / Right Panel of VIP-157
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SIP Analog Telephone Adapter
Front Panel of VIP-157S
Left / Right Panel of VIP-157S
LED Display & Button
1
PC
RJ-45 connector, to maintain the existing network structure,
connected directly to the PC through straight CAT-5 cable
RJ-45 connector, for Internet access, connected directly to
Switch/Hub through straight CAT-5 cable.
2
LAN
The LAN interface also can be connected with 802.3af PoE
switch or converter for power supply (VIP-156PE)
3
12V DC
12V DC Power input outlet
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SIP Analog Telephone Adapter
4
Reset
Reset to the factory default setting
Machine default IP is http://192.168.0.1. Press RESET button on rear panel for
over 5 seconds to reset the VoIP Phone Adapter to factory default value. (Except
speed dial and call forward settings)
LED display of VIP-156 / VIP-156PE
LED Indicators
Descriptions
PWR
Power is supplied to the device.
STATUS
The Status LED will flash when the machine is operational
LNK/ACT
OFF: the device is connected to LAN at 10Mb/s.
ON: the device is connected to LAN at 100Mb/s.
RING
OFF: the phone is idle.
ON: the phone is in use (off hook).
Blinking: the phone is ringing.
LED display of VIP-157 / VIP-157S
LED Indicators
Descriptions
STATUS
The Status LED will flash when the machine is operational
LNK/ACT
OFF: the device is connected to LAN at 10Mb/s.
ON: the device is connected to LAN at 100Mb/s.
RING 1
OFF: the phone is idle.
ON: the phone is in use (off hook).
Blinking: the phone is ringing.
RING 2
OFF: the phone is idle.
ON: the phone is in use (off hook).
Blinking: the phone is ringing.
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SIP Analog Telephone Adapter
Chapter 2
Preparations & Installation
2
Physical Installation Requirements
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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SIP Analog Telephone Adapter
Please locate your PC in the same network segment (192.168.0.x) of ATA. If you’re
not familiar with TCP/IP, please refer to related chapter on user’s manual CD or
consult your network administrator for proper network configurations
LAN IP address configuration via web configuration interface
Execute your web browser, and insert the IP address (default: 192.168.0.1) of VIP in the address bar.
After logging on machine with username/password (default: root / no password), browse “Network” -->
“Network Settings” configuration menu:
Parameter Description
IP address
LAN IP address of ATA
Default: 192.168.0.1
Subnet Mask
LAN mask of ATA
Default: 255.255.255.0
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.
L Hint
When you want to run the setup or the start function, it
must unlock the protect function #190# before setting up
network settings and ATA function via keypad.
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SIP Analog Telephone Adapter
IVR Menu Choice
Machine operation
Parameter(s)
Notes:
ATA will change to DHCP
#111#
#112xxx*xxx*xxx*
Set DHCP client
None
Client
Setup Static IP Address
xxx#
Use the * (star) key
DHCP will be disabled and
when entering a decimal
system will change to the
point.
Static IP type.
Use the * (star) key
#113xxx*xxx*xxx*
Set Network Mask
xxx#
when entering a decimal
Must set Static IP first.
point.
Use the * (star) key
#114xxx*xxx*xxx*
Set Gateway IP Address
xxx#
when entering a decimal
Must set Static IP first.
point.
Use the * (star) key
#115xxx*xxx*xxx*
Set Primary DNS Server
xxx#
when entering a decimal
Must set Static IP first.
point.
Must unlock the protect
function before setting up
#190#
Unlock
None
network settings and ATA
function via keypad.
The system will be lock
#191#
Lock
None
and can’t set up network
settings via keypad.
#195#
The system will reboot
Reboot
None
automatically.
The system will be reset to
#198#
Factory Reset
None
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.
IVR Menu Choice
Machine operation
Notes:
IVR will announce the current PC-port IP
#120#
Check PC IP Address
#121#
Check network connection type
#122#
Check the Phone Number
#123#
Check Network Mask
address of the ATA.
IVR will announce if DHCP is enabled or
disabled.
IVR will announce current enabled VoIP
number.
IVR will announce the current network mask
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SIP Analog Telephone Adapter
of the ATA.
IVR will announce the current gateway IP
#124#
Check Gateway IP Address
#125#
Check Primary DNS Server Setting
#126#
Check LAN IP Address
#128#
Check Firmware Version
address of the ATA.
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.
The following keypad commands can be used to set up the main function .
IVR Menu Choice
Machine operation
Parameter(s)
Notes:
#138#
Enable call waiting
None
Enable Call waiting
#139#
Disable call waiting
None
Disable Call waiting
#160#
Update firmware
None
Update firmware
#510#
Blind Transfer
ATA Bland Transfer
#511#
Attendant Transfer
ATA Attendant Transfer
#512#
3-way calling
ATA 3-way calling
#514#
IP transfer to PSTN
ATA transfer IP call to
PSTN side
1:G.711 u-Law, 2: G.711
a-Law, 3: G.723.1, 4:
You can set the codec you
#130+[1~8]#
Set Codec
G.729a, 5: G.726 16K, 6:
want to the first priority.
G.726 24K, 7: G.726
32K, 8: G.726 40K,
You can set the Handset
Handset Gain from
#131+[00~15]#
Set Handset Gain
gain to proper value,
0~15
default is 10
You can set the Handset
Handset Volume from
#132+[00~12]#
Set Handset Volume
volume to proper value,
0~12
default is 10
#135xxx*xxx*xxx*
xxx#
#136xxx*xxx*xxx*
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
TFTP Server IP Address
FTP Server IP Address
xxx#
You can set the Auto
0: Disable, 1: TFTP
#137+[0~2]#
Auto config mode
configuration mode, 0:
mode, 2: FTP mode
Disable, 1: use TFTP
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SIP Analog Telephone Adapter
Server, 2: user FTP Server
#145#
Forward function disable
Disable forwrad funciton
enable forward to FXS
Eanble forward to FXS
Port
Port
enable forward to FXO
Eanble forward to FXO
Port
Port
#116#
Enable PPTP function
None
Enable PPTP function
#117#
Disable PPTP function
None
Disable PPTP function
#118#
Enable VLAN function
None
Enable VLAN function
#119#
Disable VLAN function
None
Disable VLAN function
#146+Number#
#147+Number#
L 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
In most circumstances, it is no need to configure the DHCP settings.
DHCP client
The ISP will assign IP Address, and related information.
PPPoE
The ISP will assign PPPoE username / password for Internet access,
L 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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SIP Analog Telephone Adapter
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SIP Analog Telephone Adapter
Chapter 3
Network Service Configurations
3
Configuring and monitoring your ATA from web browser
The ATA integrates a web-based graphical user interface that can cover most configurations and
machine status monitoring. Via standard web browser, you can configure and check machine status
from anywhere around the world.
Overview of the web interface of ATA
With web graphical user interface, you may have:
Š More comprehensive setting feels than traditional command line interface.
Š Provides user with input data fields, check boxes, and changing machine configuration settings
Š Displays machine running configuration
To start ATA web configuration, you must have one of these web browsers installed on computer for
management
Š Microsoft Internet Explorer 6.0.0 or higher with Java support
Manipulation of ATA via web browser
Log on ATA via web browser
After TCP/IP configurations on your PC, you may now open your web browser, and input
http://192.168.0.1 to logon Phone Adapter web configuration page.
Phone Adapter will prompt for logon username/password: root / null (no password)
ATA log in page
When users login the web page, users can see the Phone Adapter system information like firmware
version, company, etc on this main page.
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SIP Analog Telephone Adapter
VoIP Phone Adapter main page
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SIP Analog Telephone Adapter
Chapter 4
VoIP Telephone Adapter Configurations
4
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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SIP Analog Telephone Adapter
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
Description
The default is Page 1. It can select Page1 ~ Page 7 to
look through Phone Book records.
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SIP Analog Telephone Adapter
Phone
Name
URL
The record number is from 1 ~ 140; it can set up 140
records in total.
The name of Phone Book records; it only can input
numerals character.
Fill in the outgoing number (Line Number) or IP
address.
Delete
Clean this item’s data.
Export csv
Save the phone book data as CSV file.
Upload
Upload the phone book file
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 “Delete” button.
Press “Reset” to erase the data that you didn’t save.
For Example:
Ex_1:
ATA has added the above phone numbers. User picks up the handset and dial “301” to make the
P2P call ([email protected]).
Ex_2:
User picks up the handset and dial “206” to make the Proxy call (17476433364).
Ex_3:
User picks up the handset and dial “202” to make the P2P call (192.168.1.2:5062).
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SIP Analog Telephone Adapter
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.
Call Forward function for VIP-156/VIP-156PT/VIP-157S
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SIP Analog Telephone Adapter
All to PSTN/ No Answer to PSTN (VIP-157): the VIP-157 not only supports 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 the URL/Number field.
IP Line Forward function for VIP-157
The IP Line Forward function’s incoming call is IP call type, and the destination is IP or PSTN call type.
The FXO Line Forward function’s 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 fixable applications.
[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 disable this feature.
Block Period: Set a time period and the phone will be blocked during the time period. If the “From” time
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SIP Analog Telephone Adapter
is larger than the “To” time, the Block time will from Day 1 to Day 2.
When you finish the setting, please click the Submit button.
[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 ringed up every day.
Alarm Type: The default is Off. If set up as On, the telephone will ring up 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)
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.
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.
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SIP Analog Telephone Adapter
Volume Settings for VIP-156T/VIP-156PT
Besides the above settings, VIP-157 also can set the volume of PSTN.
PSTN-Out Volume is to set the PSTN volume you can hear.
PSTN-In Gain is to set the volume sent out to the other side’s handset.
Volume Settings for VIP-157
Dial Plan Setting
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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SIP Analog Telephone Adapter
Dial Plan Settings for VIP-156
For VIP-157, have four more items.
Field
Description
Drop Prefix
The rule of add or replace code. If setup as Disable, it will add the
prefix number prior to the identification number. If setup as Enable, it
will replace the identification number.
Prefix
The prefix number. It only accepts the numeral and the max length is
8.
Rule Rule
The identification number. It can accept the numeral or symbol and
the max length is 40.
-
Symbol: It only accepts the [+], [x]
-
+: It means as “or”. For example, [123+456+334+5xx] even if
[123 or 456 or 334 or 5xx]
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SIP Analog Telephone Adapter
-
x: It is equal to 0~9. For example, [5xx] even if the number
begins with 5.
Dial Now rule
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.
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.
Realm 1/2/3/4/5
These options can define the switching code for each Realm No.
Area Code
Inter Digit
Stop dialing after seconds then send dialed number out.
Time(Auto Dial
Time)
Key as send #
If setup as Yes, the system will stop to receive the dialed number
when receiving the [#] key. The system also will determine the Auto
Dial Time, It will carry out the calling if there is no dialing after the Auto
Dial Time.If setup as No, 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].
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SIP Analog Telephone Adapter
Example_2: Drop prefix: Enable, Prefix: 006, Rule: 002+003+004+005+007+009
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+77xxxx”, 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.
[For VIP-157 only]
Field
Description
Auto PSTN backup
Default is Disable, it’s for PSTN backup function, when it “Enable”, if
SIP regsiter unsuccessful, it will automatically switch to PSTN line to
dial out.
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SIP Analog Telephone Adapter
To enable this function, make sure that the PSTN line is
connected to the PSTN port already.
PSTN feature Code
Default is 0*, the code for manually switching to PSTN line, and dial
out from PSTN, it can only accept the numeric and *or #, the digital
max length is 7.
Routing Type
Default is Disable, it defines the dialing route, according the [Routing
Rule] to define the dialing route which is [IP or FXO].
Routing Rule
Define the outgong rule. It can also Add/ Drop profix number, if you
want to increase more then one Routing rule, you can use “+” to
accept it by only pressing the numeric or D
D: drop
Example_5: Routing: Routing Type: FXO, Routing Rule: D007+009+0800
1. If the dialing number is “0800024365”, it will match the routing rule [0800], then system will
automatically dial out from the [FXO].
2. If the dialing number is “00986123456”, it will match the routing rule [009], then system will
automatically dial out from the [FXO].
3. If the dialing number is “00782280220”, it will match the routing rule [D007], then system will
decrease the [007] then dial out from the [FXO].The real dialing number is [82280220].
General
This page defines the volume, auto answer, Caller ID, and call waiting caller ID(CID type II),
VIP-156
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SIP Analog Telephone Adapter
VIP-157
Field
Description
Call Waiting
Default is enable.
When you are talking with other people, You can choose If you want to
hear the notice 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 notice in your current call. If you set the function to Off, then
you will not hear any notice.
Ring Timeout
Default is 60(sec). After how long the system will reply the busy(486
busy) message.
Caller ID Scheme
Set the caller ID mode, it supports FSK Bellcore, DTFM, CID-Japnan,
DTMF-Brazil, DTMF-Denmark.
FSK Bellcore: FSK caller ID mode.
DTMF: Before first ringing,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
CID Type II
To enable the show caller ID function in call waiting.
When enable this function system receives a new call in call waiting. It
will display the caller ID
Your Phone must also support CID Type2.
T.38 (FAX)
Enable/Disable T.38 FAX function.
T.38 Pass-through
Define the T.38 pass-through codec, it can support G.711 u-law/G.711
codec
a-law.
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SIP Analog Telephone Adapter
Auto Answer and PIN (VIP-157)
Field
Description
Auto Answer Type
Auto Answer: There are different incoming call types for flexible
applications. The Trunk Gateway function needs to arrange with the
registered Server System. The 3-Party subscribers could make
Off-Net call (PSTN) through the FXO port of VIP-157.
AutoAnswer
Auto Answer Counter is to set after the ring count met the number you
cournter
set then the auto answer will enable.
PIN Code
For security issue, You’d better to set the PIN Code. If you have set
PIN Code Number
the PIN code, you will hear a tone to inform you to input the PIN Code.
Then you can dial out. Please note that the PIN Code function couldn’t
function with Trunk Gateway function together.
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SIP Analog Telephone Adapter
Chapter 5
Network
5
Network Setting
This page defines the LAN setting on this page.
Field
Description
WAN Active
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
IP Address
The IP address
Subnet Mask
The sub net address
Default Gateway
The default gateway address
DNS Active
Static/ Automatically, manually set up the DNS server or automatically
accept the DNS server.
Primary DNS
The default is 168.95.192.1, it could set up the first DNS server
address.
Second DNS
The default is 168.95.1.1, it could set up the second DNS server
address.
MAC Address
The MAC of LAN port
System Name
The product model
PPPoE User Name
The PPPoE connection account name. It could input numeral or
character, the maximum date length is 63.
PPPoE Password
The PPPoE connection account password. It could input numeral or
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SIP Analog Telephone Adapter
character, the maximum date length is 63.
PPPoEService name
PPPoE Service provider name
PPPoE AC Name
PPPoE AC name.
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 web site
www.planetddns.com)
DDNS Settings for VIP-156/VIP-156PE/VIP-157
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. When you finish the setting, please click the Submit button.
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SIP Analog Telephone Adapter
Field
Description
VLAN Activity
If set up as On, it could receive VLAN messages.
VID (802.1Q/TAG)
Dispose VLAN ID is add a Tag header after enabling the VLAN
function. The realized voice packets transfer is similar with that of
VLAN. The prerequisite is it must be the same as VLAN of upper
switch. The value range is 2~4094.
User Priority
To set up the user priority.
(802.1P)
Field
Description
SIP VID
Set the SIP VLAN ID, this is an independent function that doesn’t
need to enable [VLAN Packets: Enable].
SIP User Priority
Set up the SIP Priority.
(802.1P)
RTP VID
Set the SIP RTP VID, this is an independent function that doesn’t
need to enable [VLAN Packets: Enable].
RTP User Priority
Set up the RTP Priority.
(802.1P)
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.
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SIP Analog Telephone Adapter
Caution: VIP-156/VIP-157 VPN can’t use the encryption or compression for VPN connection.
IPV6 Setting
This page defines the IPV6 setting on this page. You can program the IPV6 information.
Field
Description
IPV6 Activity
Support three IPV6 types: Auto, Fixed IPV6, IPV6 in IPV4 Tunnel
IPV6 address
Setting the WAN IPV6 address or display it.(64 bits)
SubnetPrefix Length
Default is 64
Default Gateway
IPV6 gateway address(64 bits)
LAN IPv6 Address:
IP V6 LAN address. (64 bits)
LAN IPv6 Link-Local
Link local address information.
Address
Autoconfiguration
It supports Statless, stateful(DHCP V6).
Type
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SIP Analog Telephone Adapter
Chapter 6
NAT Trans
6
Stun Setting
This page defines the STUN Enable/Disable and STUN Server IP address in this page. This function
can help your Phone Adapter work properly behind NAT. To change these settings, please follow your
ISP information. When you finish the setting, please click the Submit button.
PC Setting
This page defines the PC setting on this page.
Field
Description
Device Activity
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
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SIP Analog Telephone Adapter
address.
PC IP address
The IP address of PC port. (In the Birdge mode, the Default IP:
192.168.0.1
PC MAC Address
The MAC of PC port
Enable DHCP Server
It will allot the IP address automatically when enable this function.
IP Address
The range for DHCP IP address.
Lease Time
DHCP server lease time
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 send 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 ISP will limit the MAC address for PPPoE dial-up connection.
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.
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SIP Analog Telephone Adapter
Field
Description
Index
The serial number. There are totally 12 records from Num 1 to 12.
Activity
The activity status. The default is Disable, this record will activate if
enable.
Protocol
The TCP or UDP communication protocol.
Internal Port
For corresponding the internal port.
External Port
For corresponding the external port.
Server IP
To input the Server IP address.
Delete
Delete this item
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SIP Analog Telephone Adapter
Chapter 7
SIP Setting
7
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
Description
Realm
Which line you want to use.
Realm Activity
First you need to click Active to enable the Service Domain, then you
can input the following items.
Display Name
The serial number. There are totally 24 records from Num 0 to 23.
Phone number
The activate status. The default is Disable, this record will activate if
enable.
Authentication ID
you need to input the Register Password obtained from your ISP.
Authentication
you need to input the Register Name obtained from your ISP.
Password
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 to provide the information. Then you can skip this item.
Subscribe for MWI
Setting MWI(message-waiting indicator) function, when enable
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SIP Analog Telephone Adapter
system, it will frequently send the MWI message.
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.
You can see the Register Status on the Status page. 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 follow the steps below to register to the other ISP.
When you finish the setting, please click the Submit button.
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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SIP Analog Telephone Adapter
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.
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SIP Analog Telephone Adapter
Field
Description
SIP Expire Time
To set up the registration interval time.
SIP Expire Time
Default is General; Register interval time setting. Provide items like
Type
General (standard), 1/2, 2/3, 3/4, 4/5, 5/6, 6/7, 7/8, 8/9, 9/10。
Register server need supports 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.
2/3: expiry time * 2/3.
3/4: expiry time * 3/4.
4/5: expiry time * 4/5.
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SIP Analog Telephone Adapter
5/6: expiry time * 5/6.
6/7: expiry time * 6/7.
7/8: expiry time * 7/8.
8/9: expiry time * 8/9.
9/10: expiry time * 9/10.
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.
Register server need supports this function
SIP session timer T2
Setting the maximum retransmit interval for non-INVITE requests and
INVITE responses.
Register server need supports this function
SIP session timer
Setting the maximum retransmit interval for non-INVITE requests and
Timer B, F, H
INVITE responses。。
Register server need supports this function
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.
RPort
To change this setting, please follow your ISP information. When you
finish the setting, please click the Submit button.
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SIP Analog Telephone Adapter
Register server need supports this function
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
IPV6 RTP traffic class
Class(IPV6)
SIP Traffic
IPV6 SIP traffic class
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
To set up the interval time for transporting packets.
Jitter Buffer
To set up the size for jitter buffer packets.
SIP Server Type
Provides a different register server: General, Asterisk, BroadWorks,
Nortel, Xener, Vodtel, SKTelink, for different server systems will adjust
some system parameters
Register server need supports this function
Use user = phone
When sending the register package, in package Header will add
(Register):
the ”user=phone” message。
Register server need supports this function
Use user = phone
When sending the dialing package, in package Header will add
(Invite):
the ”user=phone” message。
Register server need supports this function
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SIP Analog Telephone Adapter
Send SIP PRACK to
When sending the SIP package, in package Header will add
Proxy:
the ”PRACK” message。
Register server need supports this function
Only Accept Trusted
Only accept call from proxy, if system receives the IP dialing, system
Certificates:
will refuse the call.
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SIP Analog Telephone Adapter
Chapter 8
Advance Setting
8
Status Log
Display and save systems running status message data. Press “Get Status Log” to back up the status
log file.
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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SIP Analog Telephone Adapter
Management-Advanced Setting
This page defines the advanced functions. When you finish the setting, please click the Submit button.
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SIP Analog Telephone Adapter
Field
Description
ICMP Not Echo
This function can disable echo when someone pings this device. It
can avoid hacker trying to attack the device
Anonymous Call
If enable this function, machine will to start the calling hidden
function, and it will not send the related Caller information.
Register server need supports this function.
Management form
When [Enable] allow user login from WAN.
WAN
Stop Feature Tone
When [Enable] if system sets the function like [Subscribe for MWI,
forward, DND], when user picks up the phone, he will hear the remind
tone [Do Do Do]
Billing Signal
There are three billing types: Polarity Reversal, Tone_12K and
Tone_16K.
Register server need supports this function.
CPC Delay
When receiving the disconnected signal, machine will cut the voltage
down to 0V after this time
CPC Duration
When starting to cut the voltage down to 0V, machine will continue this
state by this time.
IP Dialing Format
Setting IP dialing format, when [Disable] can’t use IP dialing to make
call.
Send Flash event
There are two flash formats: DTMF Event and SIP Info.
Encrypt Type
There
are
seven
encrypt
formats:
Disable,
INFINET,
AVS,
WALKERSUN1, WALKERSUN2, CSF1, CSF2, GX, VGX, RC4,
VOS_R, VGCP。
Registered server need supports this function.
Encrypt Key
Some encrypt type must enter the Encrypt Key
Registered server need supports this function.
PPPoE Retry Period
If PPPoE dial-up connection fails, machine will retry the dial-up motion
after this time.
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SIP Analog Telephone Adapter
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
PSTN Silence
Define the MAX silence time for FXO port. After the time, it will
Timeout
disconnect the line.
PSTN CID forward
It must work with [Phone – General] [Auto Answer] function or
[Phone – Caller Service]
[Forward] function。 When enable this
function, the caller ID from FXO can transfer to other devices.
Generate Flash
FXO flash time defines would you hold or hang on 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 on.
(Max)
Flash Hook Time
Minimum Flash time , to detect the call on hold or hang on.
(Min)
NET Bandwidth
Setting the limitation for LAN Bandwidth
Limit
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 Tone. When you finish the setting, please click the Submit
button.
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SIP Analog Telephone Adapter
TR-069
On this page you can program the TR-069 setting.
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
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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SIP Analog Telephone Adapter
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SIP Analog Telephone Adapter
Chapter 9
Other Setting
9
System Authority
In System Authority it can change admin/System/User login password.
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 update 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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SIP Analog Telephone Adapter
For technological consideration, we strongly suggest you refer to the following
upgrade methods for updating your device.
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)
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
There are TFTP/ FTP and HTTP to provide the auto upgrade
function.
TFTP Server
Input the TFTP Server address, and it could input the IP or Domain
Name form.
TFTP File Path
Set up the file path.
HTTP Server
Input the HTTP Server address, and it could input the IP or Domain
Name form.
HTTP File Path
Set up the file path.
FTP Server
Input the FTP Server address, and it could input the IP or Domain
Name form.
FTP Username
The login username.
FTP Password
The login password
FTP File Path
Set up the file path.
Check new firmware
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
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SIP Analog Telephone Adapter
time.
- Scheduling: The machine will follow the scheduled date
and time to check the new firmware.
Scheduling (Date)
The machine will check the new firmware between the time range
by random.
Automatic Update
There are Notify only and Automatic ways to update.
- Notify only: If there are new firmware, the ATA will send the
“Be Be Be” sound when picking up the handset to prompt
there is new firmware.
- Automatic: The device will carry out firmware update
automatically.
Firmware File Prefix
It will check the information of model name.
Next update time
It will show the next check date and time.
If the Check new firmware field selected to Power On, the machine will check the new
firmware according the scheduled time/date and power on. If there is new firmware, it
can be 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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SIP Analog Telephone Adapter
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.
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 effect.
Logout
Lougout the system, it will return to login page.
58
SIP Analog Telephone Adapter
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, 192.168.0.2
Analog telephone sets are connected to the phone (RJ-11) port of ATAs respectively
192.168.0.2
192.168.0.1
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
59
SIP Analog Telephone Adapter
Case 2: (Peer-to-Peer mode) VIP-157S Port 1 to Port 2 communications
Supposing one VIP-157S connects to two telephones, just pick up phone 1 and dial
‘192*168*0*1**5062’, phone 2 will ring.
Analog telephone sets are connected to the phone (RJ-11) ports of VIP-157S respectively
192.168.0.1
1001
1
9
2
*
1
1002
6
8
*
0
*
1
*
*
5
0
6
2
#
Test the scenario:
1. Pick up the telephone set on VIP-157S port 1, and you should be able to hear the dial-tone
2. 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
y
y
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 “#”.
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
[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 register to IPX-1980 and VIP-156_A has set Call Forward
function to VIP-156_B.
60
SIP Analog Telephone Adapter
Machine configuration on the VIP-156:
Please log in 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:
Test the scenario:
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 one VIP-157 and two VIP-156 register to IPX-1980. The VIP-157_A has set
Call Forward function to phone number 1111-2222 (PSTN).
61
SIP Analog Telephone Adapter
Machine configuration on the VIP-157:
Please log in VIP-157_A via web browser and browse the Phone Settings menu and select
the Call service config menu. On the setting page, please select the All Forward function to
PSTN choice and fill in the Forward Type and Forward Number of PSTN Phone Number
11112222, then the sample configuration screen is shown below:
Test the scenario:
1. VIP-156_C pick up the telephone
2. Dial the number 1001(VIP-157_A)
3. Because VIP-157_A has set up All Forward function to the PSTN Phone Number 11112222
4. The PSTN Phone Number 11112222 will ring up; then it picks up the telephone and
communication with the number 3003(VIP-156_C)
Case 5: Call Forward Feature_Example 3
For this example, there are one VIP-157 and two VIP-156 register to IPX-1980. The VIP-157_A has set
Call Forward function to number 2002 (VIP-156_B).
62
SIP Analog Telephone Adapter
Machine configuration on the VIP-157:
Please log in VIP-157_A via web browser and browse the Phone Settings menu and select
the Call service config menu. On the setting page, please select the All Forward function to
IP choice and fill in the Forward Type and Forward Number of of VIP-156_B, and then the
sample configuration screen is shown below:
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 picks up the telephone and communication
with the PSTN Phone Number 11112222
Case 6: Call Forward Feature_Example 4
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.
63
SIP Analog Telephone Adapter
Machine configuration on the VIP-156:
Please log in 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:
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 has 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 picks up the telephone and
communication with the VIP-156_C
Case 7: Auto Answer Feature_IP to PSTN
For this example, there are one VIP-157 and two VIP-156 and connect with Peer to Peer mode. The
VIP-157_A has set Auto Answer function for forwarding calls to arbitrary telephone. If there are
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.
64
SIP Analog Telephone Adapter
Machine configuration on the VIP-157:
STEP 1:
Please log in VIP-157_A via web browser and browse the Phone Settings menu and select
the Call service config menu. On 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 and browse the Phone Settings / General setting
menu and select the Auto Answer config menu. On 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 and then input the password 123#
5. VIP-156_C will hear the dial tone from PSTN line and then input Phone Number 11112222
65
SIP Analog Telephone Adapter
6. The Phone Number 11112222 will ring up; then it picks up the telephone and communication
with the VIP-156_C
Case 8: Auto Answer Feature_PSTN to IP
For this example, there are one VIP-157 and two VIP-156 and connect with Peer to Peer mode. The
VIP-157_A has set Auto Answer function for forwarding to arbitrary telephone. If there are incoming
PSTN calls and VIP-157_A doesn’t answer the incoming calls after specific time, the caller will hear
prompt sounds to input the password and then dial out an arbitrary IP telephone.
Machine configuration on the VIP-157:
STEP 1:
Please log in VIP-157_A via web browser and browse the Phone Settings / General setting
menu and select the Auto Answer config menu. On the setting page, please enable the Auto
Answer and PIN Code Enabled function, and then the sample configuration screen is
shown below:
STEP 2:
Please log in VIP-157_A via web browser and browse the Phone Book menu and select the
Speed Dial Settings config menu. On the setting page, please add a speed dial number for
dialing to IP address 192.168.0.2 (VIP-156_B), and then the sample configuration screen is
shown below:
66
SIP Analog Telephone Adapter
Test the scenario:
1. The Phone Number 11112222 picks 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 and then input the
password 123#
5. The Phone Number 11112222 will hear the dial tone and then input 0#
6. The IP address 192.168.0.2 (VIP-156_B) will ring up and then it picks up the telephone and
communication with the Phone Number 11112222
67
SIP Analog Telephone Adapter
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 carries out the transfer function (Press “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 calls to 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 Conference
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“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 calls to A and A will hear the prompt sounds.
3. A presses “Hold” button to hold the conversation with B, and switch to have conversation with
C.
68
SIP Analog Telephone Adapter
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.
69
SIP Analog Telephone Adapter
Appendix C VIP-156/VIP-156PE/VIP-157/VIP-157S Specifications
Product
Model
Hardware
LAN
PC
FXS (for telephone set
connection)
FXO (PSTN connection)
Protocols and Standard
Standard
Voice codec
Fax support
Voice Standard
Protocols
Network and Configuration
Access Mode
Management
Dimensions (W x D x H)
Operating Environment
Power Requirements
EMC/EMI
SIP Analog Telephone Adapter
VIP-156
VIP-156PE
VIP-157
VIP-157S
1 x 10/100Mbps RJ-45 port (802.3af PoE for VIP-156PE)
1 x 10/100Mbps RJ-45 port
1 x RJ-11
---
2 x RJ-11
1 x RJ-11
---
SIP 2.0 (RFC3261)
G.711a/u, G.723.1 (6.3k/5.3k), G.726, G.729A, G.729B, GSM
T.38
Voice activity detection (VAD)
Comfort noise generation (CNG)
Acoustic echo canceller (AEC)
G.165: Line echo canceller (LEC)
Jitter Buffer
SIP 2.0 (RFC-3261), TCP//IP, UDP/RTP/RTCP, HTTP, ICMP, ARP, DNS,
DHCP, NTP/SNTP, PPP, PPPoE
Static IP, PPPoE, DHCP
Web, keypad
94 x 72 x 30 mm
0~40 degrees C, 10~95% humidity
12V DC
CE, FCC Class B
70
SIP Analog Telephone Adapter
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
71
EC Declaration of Conformity
For the following equipment:
*Type of Product
*Model Number
: SIP Telephone Adapter
: VIP-156
* Produced by:
Manufacturer‘s Name :
Manufacturer‘s Address:
Planet Technology Corp.
11F, No 96, Min Chuan Road
Hsin Tien, Taipei, Taiwan, R. O.C.
is herewith confirmed to comply with the requirements set out in the Council Directive on the
Approximation of the Laws of the Member States relating to 1999/5/EC R&TTE.
For the evaluation regarding the R&TTE, the following standards were applied:
Emission
Conducted / Radiated
Harmonic
Flicker
Immunity
ESD
RS
EFT/ Burst
Surge Test
CS
Magnetic Field
Voltage Disp
Safety
EN 55022
EN 61000-3-2
EN 61000-3-3
EN 55024
EN 61000-4-2
EN 61000-4-3
EN 61000-4-4
EN 61000-4-5
EN 61000-4-6
EN 61000-4-8
EN 61000-4-11
EN 60950 3rd
(1998 + A1:2000 Class B)
(1995 Class A)
(1995)
(1998 + A1:2001)
(1995)
(1995)
(1995)
(1995)
(1996)
(1993)
(1994)
(2000)
Responsible for marking this declaration if the:
⌧ Manufacturer
Authorized representative established within the EU
Authorized representative established within the EU (if applicable):
Company Name:
Planet Technology Corp.
Company Address:
11F, No.96, Min Chuan Road, Hsin Tien, Taipei, Taiwan, R.O.C
Person responsible for making this declaration
Name, Surname
Jimmy Lin
Position / Title :
Product Manager
Taiwan
Place
7th July, 2005
Date
Legal Singnature
PLANET TECHNOLOGY CORPORATION
e-mail: [email protected]
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
EC Declaration of Conformity
For the following equipment:
*Type of Product
*Model Number
: PoE SIP Telephone Adapter
: VIP-156PE
* Produced by:
Manufacturer‘s Name :
Manufacturer‘s Address:
Planet Technology Corp.
11F, No 96, Min Chuan Road
Hsin Tien, Taipei, Taiwan, R. O.C.
is herewith confirmed to comply with the requirements set out in the Council Directive on the
Approximation of the Laws of the Member States relating to 1999/5/EC R&TTE.
For the evaluation regarding the R&TTE, the following standards were applied:
Emission
Conducted / Radiated
Harmonic
Flicker
Immunity
ESD
RS
EFT/ Burst
Surge Test
CS
Magnetic Field
Voltage Disp
Safety
EN 55022
EN 61000-3-2
EN 61000-3-3
EN 55024
EN 61000-4-2
EN 61000-4-3
EN 61000-4-4
EN 61000-4-5
EN 61000-4-6
EN 61000-4-8
EN 61000-4-11
EN 60950 3rd
(1998 + A1:2000 Class B)
(1995 Class A)
(1995)
(1998 + A1:2001)
(1995)
(1995)
(1995)
(1995)
(1996)
(1993)
(1994)
(2000)
Responsible for marking this declaration if the:
⌧ Manufacturer
Authorized representative established within the EU
Authorized representative established within the EU (if applicable):
Company Name:
Planet Technology Corp.
Company Address:
11F, No.96, Min Chuan Road, Hsin Tien, Taipei, Taiwan, R.O.C
Person responsible for making this declaration
Name, Surname
Jimmy Lin
Position / Title :
Product Manager
Taiwan
Place
7th July, 2005
Date
Legal Singnature
PLANET TECHNOLOGY CORPORATION
e-mail: [email protected]
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
EC Declaration of Conformity
For the following equipment:
*Type of Product
*Model Number
: VoIP Analog Telephone Adapter (1*FXS + 1*FXO)
: VIP-157
* Produced by:
Manufacturer‘s Name :
Manufacturer‘s Address:
Planet Technology Corp.
11F, No 96, Min Chuan Road
Hsin Tien, Taipei, Taiwan, R. O.C.
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
Harmonic
Flicker
Immunity
ESD
RS
EFT/ Burst
Surge Test
CS
Magnetic Field
Voltage Disp
EN 55022
EN 61000-3-2
EN 61000-3-3
EN 55024
EN 61000-4-2
EN 61000-4-3
EN 61000-4-4
EN 61000-4-5
EN 61000-4-6
EN 61000-4-8
EN 61000-4-11
(1998 + A1:2000 + A2:2003)
(2000)
(1995 + A1:2001)
(1998 + A1:2001)
(1995 + A1:1998 + A2:2000)
(2002 + A1:2002)
(1995 + A1:2000 + A2:2001)
(1995 + A1:2000)
(1996 + A1:2000)
(1993 + A1:2000)
(1994 + A1:2000)
Responsible for marking this declaration if the:
⌧ Manufacturer
Authorized representative established within the EU
Authorized representative established within the EU (if applicable):
Company Name:
Planet Technology Corp.
Company Address:
11F, No.96, Min Chuan Road, Hsin Tien, Taipei, Taiwan, R.O.C
Person responsible for making this declaration
Name, Surname
Jimmy Lin
Position / Title :
Product Manager
Taiwan
Place
16 March, 2006
Date
Legal Signature
PLANET TECHNOLOGY CORPORATION
e-mail: [email protected]
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
EC Declaration of Conformity
For the following equipment:
*Type of Product
*Model Number
: VoIP Analog Telephone Adapter (2*FXS)
: VIP-157S
* Produced by:
Manufacturer‘s Name :
Manufacturer‘s Address:
Planet Technology Corp.
11F, No 96, Min Chuan Road
Hsin Tien, Taipei, Taiwan, R. O.C.
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
Harmonic
Flicker
Immunity
ESD
RS
EFT/ Burst
Surge Test
CS
Magnetic Field
Voltage Disp
EN 55022
EN 61000-3-2
EN 61000-3-3
EN 55024
EN 61000-4-2
EN 61000-4-3
EN 61000-4-4
EN 61000-4-5
EN 61000-4-6
EN 61000-4-8
EN 61000-4-11
(1998 + A1:2000)
(2000)
(1995 + A1:2001)
(1998 + A1:2001)
(1995 + A1:2001 + A2:2000)
(2002 + A1:2002)
(1995 + A1:2000 + A2:2001)
(1995 + A1:2000)
(1996 + A1:2000)
(1993 + A1:2000)
(1994 + A1:2000)
Responsible for marking this declaration if the:
⌧ Manufacturer
Authorized representative established within the EU
Authorized representative established within the EU (if applicable):
Company Name:
Planet Technology Corp.
Company Address:
11F, No.96, Min Chuan Road, Hsin Tien, Taipei, Taiwan, R.O.C
Person responsible for making this declaration
Name, Surname
Jimmy Lin
Position / Title :
Product Manager
Taiwan
Place
17 March, 2006
Date
Legal Signature
PLANET TECHNOLOGY CORPORATION
e-mail: [email protected]
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
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