TM

RingStar ZT-1000

Dual-FXS ATA
User Guide

Ver. 1.2.7A1

Zinwell Corporation
 COPYRIGHT
Copyright © 2005-2006 of Zinwell Corporation, Wen-Hua road, Hsinchu Industrial
Park, Taiwan. All rights reserved.

The contents of this publication may not be reproduced in any part or as a whole,
transcribed, stored in a retrieval system, translated into any language, or
transmitted in any form or by any means, electronic, mechanical, magnetic, optical,
chemical, photocopying, manual, or otherwise, without the prior written permission
of Zinwell Corporation.

Published by Zinwell Corporation. All rights reserved.

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 WARRANTY DISCLAIMER
Zinwell Corporation does not assume any liability arising out of the application or
use of any products, or software described herein. Neither does it convey any
license under its patent rights nor the patent rights of others.

Zinwell Corporation further reserves the right to make changes in any products
described herein without notice.

This publication is subject to change without notice.

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 TRADEMARK
“RingStar” (Zinwell Corporation VoIP family) is a registered trademark of Zinwell
Corporation.

Other trademarks mentioned in this publication are used for identification

purposes only and may be properties of their respective owners.

iii Zinwell Corporation

 SAFETY WARNING
For your safety, be sure to read and follow all warning notices and instructions:

‹ To reduce the risk of fire, use only No. 26 AWG (American Wire Gauge) or
larger telecommunication line cord.

‹ Do NOT open the device or unit. Opening or removing covers can expose
you to dangerous high voltage points or other risks. ONLY qualified service
personnel can service the device. Please contact your vendor for further
information.

‹ Use ONLY the dedicated power supply for your device. Connect the power
cord or power adaptor to the right supply voltage (110V AC in North America
or 230V AC in Europe).

‹ Do NOT use the device if the power supply is damaged as it might cause
electrocution. If the power supply is damaged, remove it from the power
outlet.

‹ Do NOT attempt to repair the power supply. Contact your local vendor to
order a new power supply.

‹ Place cables carefully so that no one will step on them or stumble over them.
Do NOT allow anything to rest on the power cord and do NOT locate the
product where anyone can walk on the power cord.

‹ Do NOT install nor use your device during a thunderstorm. There may be a
remote risk of electric shock from lightning.

‹ Do NOT expose your device to dampness, dust or corrosive liquids.

‹ Do NOT use this product near water, for example, in a wet basement or near
a swimming pool.

‹ Do NOT obstruct the device ventilation slots, as insufficient airflow may harm
your device.

‹ Do NOT store things on the device.

‹ Connect ONLY suitable accessories to the device.

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 ABOUT THIS GUIDE
Congratulations on your purchase of RingStar Dual-FXS ATA. Your RingStar
family product ZT-1000 is easy to install and configure.

This manual is designed to guide you through the configuration of your ZT-1000
for its various applications and provides background information on the ZT-1000
features.

Note: Use the web configuration environment to configure your RingStar

Dual-FXS ATA ZT-1000.

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 Table of Contents
COPYRIGHT......................................................................................................................................................... I
WARRANTY DISCLAIMER .......................................................................................................................... II
TRADEMARK.....................................................................................................................................................III
SAFETY WARNING ........................................................................................................................................ IV
ABOUT THIS GUIDE.........................................................................................................................................V

PART I INTRODUCTIONS ............................................................ 1

CH. 1 INTRODUCTION TO VOIP ..................................................................................................................2
1.1 Advantages of VoIP to Businesses.......................................................................................................2
1.2 SIP – Most Popular VoIP Protocol.....................................................................................................2
1.2.1 SIP Identities ........................................................................................................................................................ 3
1.2.2 Servers in SIP World........................................................................................................................................... 3
1.3 VoIP-Friendly Networking Concepts..................................................................................................3
1.3.1 NAT, Network Address Translation.................................................................................................................. 3
1.3.2 STUN, Simple Traversal of UDP through NAT.............................................................................................. 4
1.3.3 Voice Codec......................................................................................................................................................... 4
CH. 2 KNOWING RINGSTAR ZT-1000......................................................................................................5
2.1 Overview of RingStar Dual-FXS ATA, ZT-1000................................................................................5
2.2 Key Features of RingStar Dual-FXS ATA, ZT-1000 .........................................................................5
2.3 Outlook of RingStar Dual-FXS ATA, ZT-1000...................................................................................6
2.4 Web Configuration Environment.........................................................................................................7

PART II ATA SETTINGS ................................................................ 9

CH. 3 CONFIGURING ZT-1000.................................................................................................................10
3.0 Preparation for configuration & IVR Instructions ..........................................................................10
3.1 Dial Settings.........................................................................................................................................13
3.1.1 Phone Book........................................................................................................................................................ 13
3.1.2 Speed Dial .......................................................................................................................................................... 14
3.2 Phone Settings .....................................................................................................................................15
3.2.1 Call Forward ...................................................................................................................................................... 15
3.2.2 Volume Adjustment.......................................................................................................................................... 16
3.2.3 Call Blocks......................................................................................................................................................... 17
3.2.4 Caller ID............................................................................................................................................................. 17
3.2.5 Auto Dial............................................................................................................................................................ 18
3.2.6 Flash Time.......................................................................................................................................................... 18
3.2.7 Call Waiting ....................................................................................................................................................... 19
3.3 Network Settings..................................................................................................................................19
3.3.1 Network Status................................................................................................................................................... 19
3.3.2 Network Mode................................................................................................................................................... 20
3.3.3 SNTP .................................................................................................................................................................. 22
3.3.4 ICMP Setting ..................................................................................................................................................... 23
3.4 SIP Settings..........................................................................................................................................23
3.4.1 Service Domain ................................................................................................................................................. 23
3.4.2 Port Settings ....................................................................................................................................................... 24
3.4.3 Codec Selection ................................................................................................................................................. 24
3.4.4 DTMF................................................................................................................................................................. 25
3.4.5 QoS ..................................................................................................................................................................... 26
3.4.6 Others.................................................................................................................................................................. 26
3.5 NAT Traversal.....................................................................................................................................27
3.5.1 STUN Setting..................................................................................................................................................... 27
3.6 Others...................................................................................................................................................27
3.6.1 Auto Configuration ........................................................................................................................................... 27
3.6.2 Country Setting.................................................................................................................................................. 29
3.6.3 Call History ........................................................................................................................................................ 29
3.6.4 Dial Plan............................................................................................................................................................. 30
3.7 System Administration ........................................................................................................................30
3.7.1 System Authority............................................................................................................................................... 30
3.7.2 Firmware Upgrade............................................................................................................................................. 31
3.7.3 Revert to Default................................................................................................................................................ 31

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 3.8 Apply & Reboot...................................................................................................................................32
3.9 Logout ..................................................................................................................................................32

PART III ILLUSTRATIONS............................................................ 33

CH. 4 FAQ & TROUBLESHOOTING ...........................................................................................................34
4.1 How can I make a phone call?...........................................................................................................34
4.2 How can I pick up another call while I’m on the phone?................................................................34
4.3 How can I make a 3-way conference call?.......................................................................................34
4.4 How can I transfer a call to others?..................................................................................................34
4.5 Why can’t I login to the RingStar ZT-1000?.....................................................................................35
4.6 It seems something wrong about the LED?......................................................................................35
4.7 Why can’t I access the Internet?........................................................................................................35
4.8 Why can’t I make VoIP phone calls to others? ................................................................................35
4.9 ZT-1000 does not respond correctly to key press sometimes..........................................................35

APPENDIX...................................................................................... 36
APP. A LIST OF FIGURES ...............................................................................................................................37

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 Part I Introductions

Topics in Part I:

z Ch. 1 Introduction to VoIP

z Ch. 2 Knowing RingStar Dual-FXS ATA ZT-1000

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 Ch. 1 Introduction to VoIP
VoIP stands for Voice over Internet Protocol, which means transmitting voice over
the most widely accepted network protocol around the world. Internet Protocol
networking is widely supported by almost all kinds of networks. VoIP, however,
can also be implemented inside LAN only without Internet involved if proper LAN
environment is ready. In addition, with improvements, business class VoIP system
can integrate data, voice, and video in the same network. This takes simplified
management rules into three different systems / networks and results in enhanced
performance for all kinds of organizations.

Later on, this chapter will guide you through the basic elements of the VoIP and
some related terms involved as below:

z Advantages of VoIP to Businesses

z SIP – Most Popular VoIP Protocol
z VoIP-Friendly Networking Concepts

1.1 Advantages of VoIP to Businesses

Telephony system was invented around 100 years ago by Alexander Graham Bell.
Though telephones expedite the business communications, it’s expensive
especially for international phone calls. Internet Protocol has become the world
wide standard. With remarkable improvements in broadband access, VoIP
technology and IP Telephony, it is about time to provide businesses high quality IP
phone calls at comparably low cost with good reliability. Since voice is transmitted
over the Internet there is no demarcation of any physical boundary, this dissolves
any need of being charged at very high rate for making international phone calls.

IP Telephony focuses on using your existing telephony resources to build a vibrant

communications network that enhances productivity. A common business class
VoIP system provides all the features you’ve been familiar with—voicemail, call
waiting, call forwarding, to name a few—as well as many new exciting features
such as making LOCAL calls at whatever location you may be in the world.

Business world needs reliability and productivity as well. In conventional way,

telephony system and computer-based system are two different networks. Hence,
the two different networks require normally two times of man-power to manage,
and they are hardly managed under a set of unified managing principles. Business
class VoIP system will be able to provide easy-to-manage, business-driven
architecture at a cost that is competitive with your current expenditures.

1.2 SIP – Most Popular VoIP Protocol

SIP stands for Session Initiation Protocol which is an application-layer signaling
protocol that handles the VoIP phone calls over the Internet. Unlike other VoIP
protocols, SIP handles signaling only, and leverages other SIP-related protocols
to co-work with media sessions accordingly. The simple and flexible enough
architecture brings great chances to SIP for expanding, such as Video phone
calls.

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 Later on, we’ll introduce some terminologies that are common in the VoIP worlds,
especially SIP related. In addition, we strongly recommend the system
administrator to read other VoIP related documents, textbooks to have better
understanding in order to construct a VoIP-friendly network.

1.2.1 SIP Identities

A SIP identity (a.k.a. SIP address or SIP Account) is similar to an email address
which is used for identifying who’s who in the SIP world just like email address
used for identifying the owner. The format of a SIP identity is like an email address,
presented as SIP-Number@SIP-Service-Domain. As the expression shows, there
are two major parts in the SIP account; the SIP-Number and SIP-Service-Domain.

z The SIP number is the first part of the SIP URI that is located in front of the
“@” symbol. A SIP number can be alphabet characters just like in an e-mail
address (johndoe@ITSP-XYZ.com for example) or numeric digits like a
regular telephone number (1234567@ITSP-ABC.com for example).
z The SIP-Service-Domain is the domain name in a SIP URI. It is used to
identify who the service provider is. Normally, it will be the ITSP or the Internet
domain name which your company has.

1.2.2 Servers in SIP World

The architecture of SIP protocol is client-server based. Every SIP device or
application program will act as SIP client and SIP server as well. The SIP device
acts as client when sending out SIP requests, and acts as SIP server once
responding to the SIP requests.

The Servers we talk about here aim only at the device, and / or application
program which is used to provide VoIP services or application program which is
used by SIP-Service-Domian owner to provide its services to its subscribers.
Sometimes, the SIP Registration Server will be used for distinguishing purpose.

There are three kinds of the server:

z SIP Proxy Server; it relays call signaling, i.e. acts as both client and server
operating in a transactional manner, i.e., it keeps no session state
z SIP Redirect Server; it redirects caller to other servers
z SIP Registrar; it accepts registration requests from users and maintains
user’s whereabouts at a Location Server.

1.3 VoIP-Friendly Networking Concepts

There are many common features from regular router which will block your VoIP
communications, or make your network environment very VoIP-unfriendly. Here
are some topics which you need to take care of, or might be helpful in making a
VoIP-friendly network.

1.3.1 NAT, Network Address Translation

One of the most common features which router provides in order to provide
Internet access to all the computers in your LAN environment is “NAT”. In the
simplest case, NAT changes the source IP address of a packet received from a
device to another IP address before forwarding the packet towards its destination.
When the response comes back, NAT translates the destination address back to
the device's IP address and forwards it to the device.

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 NAT routers are commonly used to translate private (or internal) IP addresses in
packet headers to public (or external) IP addresses and vice versa. A NAT router
maps a private IP address & port; pairs it to a public IP address & port and
whenever the NAT router receives a packet with that public IP address & port, it
knows how to reroute the packet back to the private IP address & port.

The RingStar ZT-1000 must know the public IP address which you get from your
ISP in order to provide its services. If there is a NAT router between the RingStar
ZT-1000 and your broadband modem then your RingStar ZT-1000 probably has a
private IP address. In this case, the SIP session will not be able to establish
properly, and you must configure the NAT router to forward traffic with the correct
IP address & SIP port number to solve this problem.

1.3.2 STUN, Simple Traversal of UDP through NAT

STUN can be used to help your RingStar, ZT-1000, to find the present types of
NAT routers and/or firewalls between it and the public Internet. STUN also allows
the RingStar ZT-1000 to find the public IP address that NAT has assigned, so that
the RingStar ZT-1000 can embed it in the SIP data stream. However, STUN does
not work with symmetric NAT routers or firewalls. For more information, please
refer to RFC 3489 on STUN.

1.3.3 Voice Codec

A codec (coder/decoder) converts analog voice signals into digital signals and
decodes the digital signals back into voice signals. Each codec has its positive
part, good voice quality and bad part, bandwidth consumption. RingStar ZT-1000
supports the following codecs.

z ITU-T G.711; G.711 is a Pulse Code Modulation (PCM) waveform codec.

G.711 provides very good sound quality but requires 64kbps of bandwidth.
z ITU-T G.729; G.729 is an Analysis-by-Synthesis (AbS) hybrid waveform
codec that uses a filter based information about how the human vocal tract
produces sounds. G.729 provides good sound quality and reduces the
required bandwidth to 8kbps.

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 Ch. 2 Knowing RingStar ZT-1000
In this chapter, we’ll go through the following topics in order to learn more about
what RingStar Dual-FXS ATA ZT-1000 can provide, and familiarize ourselves with
RingStar ZT-1000. The topics are as follows:

z Overview of RingStar Dual-FXS ATA

z Key Features of RingStar ZT-1000
z Outlook of RingStar ZT-1000
z Web Configuration Environment

2.1 Overview of RingStar Dual-FXS ATA, ZT-1000

The RingStar Dual-FXS ATA ZT-1000 user guide provides instructions for
administering the RingStarZT-1000. RingStar ZT-1000 is a device converting
analog phones into VoIP-ready devices which support most popular VoIP protocol
- SIP (Session Initiation Protocol).

RingStar ZT-1000 can instantly convert your existing PSTN phones into
VoIP-ready devices and enjoy the inexpensive IP telephony services without
purchasing extra IP phones. It features two RJ-45 Ethernet port for sharing the
Internet accessibility with your PC simultaneously while making IP calls.

With built-in NAT function, business persons and/or travelers can use their
ZT-1000 and computer at the same time with the capability of accessing Internet
through broadband Internet service which hotels provide under single IP
architecture, especially web authentication is required.

2.2 Key Features of RingStar Dual-FXS ATA, ZT-1000

RingStar Dual-FXS ATA ZT-1000 is a state-of-the-art design featuring several

cutting-edge functions, and it provides users a small box with many things such as
combining data communication and voice communications. The key features are
shown as below.

z Supporting SIP v2 (RFC 3261) Standard

z Auto-Provisioning Function Ensures Easy Configuration with RingStar IP PBX
z Supporting Multiple Codecs
z Additional RJ-45 Port for PC Connection
z Supporting Multiple IP Assignment: PPPoE, DHCP, and Static IP
z Supporting VAD (Voice Activity Detection) and CNG (Comfortable Noise
Generation)
z Adaptive Jitter Buffer
z Acoustic Echo Cancellation
z DTMF Tone Generation

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 2.3 Outlook of RingStar Dual-FXS ATA, ZT-1000

Figure 1 Front view of RingStar Dual-FXS ATA, ZT-1000

LED Status Description

1 On ZT-1000 is receiving power.

(Power LED) Off ZT-1000 is not receiving power.
On WAN port is connected to your network environment.
2
Blinking WAN port is transmitting / receiving data.
(WAN LED)
Off WAN port is disconnected.
On LAN port is connected to your network environment.
3
Blinking LAN port is transmitting / receiving data.
(LAN/PC LED)
Off LAN port is disconnected.
On Phone 1 is registered successfully.
4a
Blinking Phone 1 has voicemail in SIP server.
(Phone 1 LED)
Off Phone 1 failed the registration.

4b On Phone 1 is off-hook.
(Phone 1 Blinking Phone 1 has incoming call.
Status)
Off Phone 1 is on-hook.
On Phone 2 is registered successfully.
5a
Blinking Phone 2 has voicemail in SIP server.
(Phone 2 LED)
Off Phone 2 failed the registration.

5b On Phone 2 is off-hook.
(Phone 2 Blinking Phone 2 has incoming call.
Status)
Off Phone 2 is on-hook.

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 Figure 2 Rear view of ZT-1000

Item Description

Power Connection of RingStar ZT-1000 power adaptor

Phone 2 Connection port to analog phone 2
Phone 1 Connection port to analog phone 1
LAN / PC For connecting to your PC
WAN For connecting to your networking environment
Reset Reset button of RingStar ZT-1000

NOTE:

If NAT function is on, do remember to connect WAN port to your networking environment and use
LAN/PC port for connecting to your PC or laptop computer. Otherwise, RingStar ZT-1000 might
interference the behavior of your LAN environment. For more information, please refer to Section
3.3.2 “Network Mode.”

2.4 Web Configuration Environment

Configuring RingStar ZT-1000 is very simple and as easy as normal networking

devices. Simply use your web browser to do the configuration. Detailed
information and configuration way will be discussed in the following sections.

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 Figure 3 Web Configuration environment of RingStar ZT-1000

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 Part II ATA Settings

Topics in Part II:

z Ch. 3 Configuring ZT-1000

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 Ch. 3 Configuring ZT-1000
In this chapter, this guide will lead you through all the functions of RingStar
ZT-1000, including the following topics:

z Preparation for configuration & IVR Instructions

z Dial Settings
z Phone Settings
z Network Settings
z SIP Settings
z NAT Traversal
z System Administration
z Others

3.0 Preparation for configuration & IVR Instructions

RingStar ZT-1000 is with a built-in function called IVR, Interactive Voice

Responses, which answers the instruction you entered though the keypad of your
phone. Before configuring your ZT-1000, there are several things your need to do
first as listed below. In addition, we suggest you to have similar network topology
like figure 4 for the configuration.

1. Make sure you have workable network environment. (Local Area Network)
2. Connecting your ZT-1000 to the network.
3. Connecting your analog phone(s) to ZT-1000.
4. Set your ZT-1000 IP address assignment type as DHCP client by entering the
IVR instruction through the keypad of your analog phone, “#111#”. (Note: The
device will reboot for 30 seconds.)
5. Use your analog phone to get the IP address of your ZT-1000 by entering the
IVR instruction through the keypad of your analog phone, “#120#”. Then you
will hear an interactive voice response (IVR) for your IP address.

Figure 4 Recommended network topology for configuring ZT-1000

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 NOTE:

1. The way of dialing phone number in the VoIP world is bit different from PSTN world. For
ZT-1000, every phone number you entered should end with pound (or hash) key “#” which
is for triggering ZT-1000 to do the dialing.

2. Since the DOT key, “.”, is not common in normal PSTN phones, ZT-1000 treats STAR key,
“＊” as DOT. Assuming you want to enter 192.168.1.1 through keypad, you need to enter
192*168*1*1.

IVR Instructions:

Command Definition Example

#111 DHCP client #111#
#112192＊168＊0＊100#
#112 Set Fixed IP
set IP as 192.168.0.100
#120 Voice IP address #120#
#121 Voice IP type #121#
#122 Voice SIP register ID #122#
#123 Voice netmask #123#
#124 Voice gateway #124#
#125 Voice DNS #125#
#128 Voice firmware version #128#
#1301#
Prioritize codec sequence as
711 u-law / 711 a-law / 729
Priority Type:
#130 + priority Prioritize Codec 1: codec priority
type Sequence G.711 u-law / G.711 a-law / G.729
2: codec priority
G.711 a-law / G.711 u-law / G.729
3: codec priority
G.729 / G.711 u-law / G.711 a-law
#13109#
#131 Handset gain
Set handset gain as 9
#13209#
#132 Handset volume
Set handset volume as 9
#138 Enable call waiting #138#
#139 Disable call waiting #139#

#140 + #1401101#
Forward type Always forward to 101
Forward setting
+ Forwarded Forward Type:
Phone No 1: Always Forward

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 2: Busy forward
3: No answer forward
#141 Disable forward setting #141#
#195 Save & reboot #195#
#198 Reset to default #198#

For configuring your ZT-1000, simply turn on your web browser, ex: Internet
Explorer, Netscape or Firefox are all OK, and enter http://<ip address>:9999.
Then, you’ll see the something like figure 5 below.

Default value for Administrator:

system

Default value for Administrator:

test

Figure 5 Login page of ZT-1000 for configuration

Since ZT-1000 is a dual-FXS ATA, it means ZT-1000 can serve two phones; in
other words, two users. During the configuration, the accessible functions you can
or can not configure depend on the username / password you entered. As
administrator, you have the full rights for all the settings; however, if you login as a
normal user, you will be stop from accessing non-authorized features, such as the
other’s phone book. The username / password of regular users are “user1 /
test”, and “user2 / test,” and they can be changed in System Authority; please
refer to Section 3.7.1.

After entering the username and password, you’ll see the main picture of ZT-1000
web configuration environment. Here you can have the basic information of
ZT-1000, shown as figure 6 below.

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 Figure 6 Main page of ZT-1000 web configuration environment

3.1 Dial Settings

The first feature set of RingStar ZT-1000 is “Dial Settings” which contains two
functions “Phone Book” and “Speed Dial.” Through configuring the functions, you
can turn ZT-1000 into a very handy companion while using your VoIP service.

3.1.1 Phone Book

In this page, you can enter the position, name, and phone number, where position
is the sequence of phone book you are about to store the contact information, as
shown in figure 7 and figure 8.

Figure 7 Page of Phone Book

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 Figure 8 Page of Phone Book (Continued)

If you want to remove certain contact(s), simply click the check box of each entry
and then click “Delete Selected” button. Please note that if you do so, the
contact(s) you select will be removed permanently. If you select “Delete All”, then
all the contact(s) from 0~59 will be removed ALL.

3.1.2 Speed Dial

The function of “Speed Dial” is a bit like shortcut. Through setting the speed dials,
you can call anyone in the speed dial list simply by dialing “*0#” to “*9#” instead
of entering the complete phone numbers.

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 Figure 9 Page of Speed Dial

In this page, you can enter the position, name, and phone number, where position
is the sequence of speed dial you are about to store the contact information, as
shown in figure 9.

If you want to remove certain contact(s), simply click the check box of each entry
and then click “Delete Selected” button. Please note that if you do so, the
contact(s) you select will be removed permanently.

3.2 Phone Settings

The feature set of “Phone Settings” contains several functions which are phone
related, including “Call Forward,” “Volume Adjustment,” “Call Blocks,” “Auto Dial,”
“Flash Time,” and “Call Waiting.”

3.2.1 Call Forward

Call Forward is used for the situations which you can’t receive the calls in front of
the phone and you don’t want to miss the calls. By setting the Call Forward, all the
incoming calls will be forwarded to the phone number you set according to the
situation.

There are three types of forwarding; “Unconditional Forward,” “Busy Forward,”

and “No Answer Forward.” “Unconditional Forward” will forward all the incoming
calls to the phone number you set immediately, and make the other two types
useless.

“Busy Forward” will be activated if you are on the phone and there’s an incoming
call. In this case, the incoming call will be forwarded to the phone number you set,
such as your colleague’s or your mobile phone.
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 “No Answer Forward” means the incoming calls will be forwarded to the phone
number you set if you can not pick up the call in time. The “time out” period
depends on what you set in the field of “time out” in this page.

Figure 10 Page of Call Forward

NOTE: Since the “Unconditional Forward” has highest priority, it will override the settings of “Busy
Forward” and “No Answer Forward.”

3.2.2 Volume Adjustment

For adjusting the volume, you can change the value of “Handset Volume” which is
for setting the volume for you can hear from the handset. And the “Handset Gain”
is to set the volume send out to the counter party’s handset.

Figure 11 Page of Volume Adjustment

If you feel the volume is too low or too loud during phone conversation, you can
log into the ZT-1000 and adjust the setting. Once the value submitted, the
changes will be applied immediately without rebooting the device.

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 3.2.3 Call Blocks

“Call Blocks,” as the name expresses that it will block the incoming calls if it
meets the criteria you set.

There are two kinds of call blocks; “Always Block” and “Periodic Block.” “Always
Block” will keep your phone absolutely silent; it means nobody can reach you
through the phone calls unless you disable this setting.

If you just don’t want to receive the phone calls in certain period, then you need to
use “Periodic Block.” By setting “Block Period” on and entering the time of “From”
and “To,” your phone will be quiet during such period.

Figure 12 Page of Call Blocks

Note:
z Similar to the function of Call Forward, if “Always Block” has been set, “Periodic Block” will be
useless even the block period has expired. Be very careful about this function, otherwise you
probably will miss a lot of phone calls.
z Remember to set “Periodic Block” off. Otherwise, the function will be activated every day
during such time interval.

3.2.4 Caller ID

The “Caller ID” is for selecting which kind of Caller ID system your phone supports.
By selecting the appropriate Caller ID type, ZT-1000 will pass the Caller ID to your
phone for display.

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 Figure 13 Page of Caller ID

Figure 14 Page of Caller ID (Continued)

3.2.5 Auto Dial

The function of “Auto Dial” means you don’t need to press “#” key for sending out
the phone number you just entered after certain period of time.

Figure 15 Page of Auto Dial

This function helps eliminate the necessity of entering “#” key for completing the
phone call, and make VoIP calls work like PSTN world; however, if the time period
is too short, ZT-1000 will send out the uncompleted number to the VoIP server
and result in failed or wrong calls. If you set this number to large, it might confuse
the user why the phone will take so long to make a phone call. We suggest you to
try normal PSTN phone calls to get the time period of failed calls due to slow
entering of phone number.

3.2.6 Flash Time

Not every analog phone has the flash key. In such case, you will be stop from
utilizing many functions, such as taking other incoming call without hanging up the
current conversation.

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 By flashing the hook of the phone set, it simulates the behavior of flash key in
advanced analog phones. The time period is used for setting how long you can
keep the hook pressed without being misjudged as hanging up the line.

Figure 16 Page of Flash Time

3.2.7 Call Waiting

The “Call Waiting” function brings you the alternative to take the other phone call
while you are on the phone already. If the Call Waiting has set, you will hear the
call waiting tone during your conversation with someone else.

If you want to pick up this incoming call without stop current conversation, simply
press the flash key your analog phone provides or using the phone hook to
simulate the flash which describe in previous section. Once this call is finished,
simply press the flash key again, you will be able to continue the call on hold.

Figure 17 Page of Call Waiting

3.3 Network Settings

The third feature set of ZT-1000 is about networking which contains “Network
Status,” “Network Mode,” and “SNTP.”

3.3.1 Network Status

This page simply shows the current network status of ZT-1000. As figure 18
shown, the information contains “IP Assignment Type,” “Current IP Address,”

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 “Netmask,” “Gateway Address,” and the IP addresses of DNS Server 1 and DNS
Server 2.

Figure 18 Page of Network Status

3.3.2 Network Mode

The Network Mode determines what ZT-1000 works within the network
environment you are in. There are two modes in this function, “Bridge mode” and
“NAT mode.”

For the “Bridge mode,” the other RJ-45 port works like normal Ethernet hub which
transparently delivers all the packets to the counter parties of connected PC and
the network environment. In this case, no matter which RJ-45 port you use to
connect to your PC, it makes no difference.

For “NAT mode,” ZT-1000 works like normal router you can buy from the market,
so you need to do some more settings. In such case, the WAN RJ-45 port should
connect to your network environment, and LAN/PC port should connect to your
PC or Ethernet hub for sharing the network accessibility. Mis-connections of RJ-45
ports might make your network environment in trouble.

To keep it simple, suggest you to follow the descriptions shown on the back panel
of ZT-1000; use WAN port to connect to your network environment and LAN/PC
port to connect to your PC. By following the rule, it won’t bring any troubles to your
network environment even your ZT-1000 has been set as working under “NAT
mode.”

3.3.2.1 Bridge Mode Setting

z Select IP Type
z Complete the TCP/IP configurations

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 Figure 19 Page of Network Mode - Bridge Mode
Fixed IP
Fill in all the information needed, such as IP address, netmask, gateway, and the
IP address of DNS Server.
DHCP Client
If IP type chosen as DHCP Client, you can specify the IP address of DNS server
you prefer or just leave it blank.
PPPoE
If you connect your RingStar ZT-1000 directly to your broadband modem, you
probably need to use PPPoE as IP type. Then, you need to enter the User Name
and Password into the fields accordingly.

Note:

Most of cases, you don’t need to take care of the MAC Clone field. If you’ve found other networking
devices can not work properly which are connected to the LAN environment through ZT-1000, then
you probably need to fill the MAC address into MAC field.

3.3.2.2 NAT Mode Setting

Under NAT mode, the ZT-1000 will act like a regular router so that there are at
least three parts of information need to be input correctly.

z Fill IP related information of LAN

z Turn on DHCP services
z Complete the WAN settings
LAN IP Address
This is the IP address of your ZT-1000 in NAT mode, so that other network device,
including your PC, can find your ZT-1000.

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 DHCP Server
This function is to decide what the range of the IP addresses will be assigned to
other network devices. Turn DHCP Server on, give the range of IP addresses, and
the lease time the IP address.

Figure 20 Page of Network Mode – NAT mode

WAN Setting
The last thing to do is to complete the WAN setting of your ZT-1000. You can refer
to “Bridge Mode” to complete this part. This function is to decide what the range of
the IP addresses will be assigned to other network devices. Turn DHCP Server on;
give the range of IP addresses, and the lease time the IP address.

3.3.3 SNTP

The SNTP protocol is for synchronizing the system time of RingStar ZT-1000 with
the time server somewhere on the Internet.

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 Figure 21 Page of SNTP Setting

For using this function, your need to enter the server address, the Time Zone and
Sync Time which means the time period of ZT-1000 synchronizes the system
time.

3.3.4 ICMP Setting

ICMP stands for Internet Control Message Protocol; herein, it’s used for deciding
whether ZT-1000 should respond to the source(s) of incoming packets for
acknowledging its status, online or not. Once the ICMP Not Echo is set to ON,
ZT-1000 will not answer any incoming ICMP packets; means ZT-1000 will behave
like it’s dead while pinging its IP address.

Figure 22 Page of ICMP Setting

3.4 SIP Settings

3.4.1 Service Domain

Service Domain is the most important part of RingStar ZT-1000 which is for
activating your VoIP service. Fill in the information you have from ITSP and enter
into fields accordingly.

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 Figure 23 Page of Service Domain

Basically, you will have the VoIP subscription information from ITSP which are
“Register Name,” “Register Password,” “Domain Server,” “Proxy Server,” and
“Register Period.”

3.4.2 Port Settings

The SIP Port is used for VoIP phone call signaling, it’s 5060 in most cases.
However, you can change the SIP port to number if you prefer.

Figure 24 Page of Port Settings

3.4.3 Codec Selection

Codec stands for Coder / Decoder which converting your voice into VoIP packets.
Through codec, it can compress the size of your voice and transmitting on the
Internet efficiently. RingStar ZT-1000 supports ITU-T standards, G.711 u-law,
G-711 a-law, and G.729. If G.711u-law or G.711 u-law is in use, the voice quality
will be very good. However, you need a plenty of bandwidth. If you find the voice is
discontinued all the time, probably the bandwidth is not enough for support G.711
u-law or a-law. In such case, please use G.729.

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 For a popular 2M/256K ADSL program, the WAN uplink speed would be 256 and
the WAN downlink speed would be 2048 since the unit is in kbps. The Uplink VoIP
reserved could be, say, 192 out of the total 256 kbps to allow 2 concurrent G.711
calls.

In addition to VAD, Voice Activity Detection, it is used for reducing the bandwidth
consumption. Normally, enabling VAD function can save around 50% traffic. Turn
on the VAD function can effectively solve the bandwidth problem.

Figure 25 Page of Codec Selection

3.4.4 DTMF

This page is for setting the DTMF, Dial Tone Multi Frequency, which is used for
sending to the SIP server in ITSP. Basically, there are three kinds of DTMF type
used in VoIP world. Choose the one your ITSP allow you to use, otherwise you
might encounter problems.

Figure 26 Page of DTMF

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 3.4.5 QoS

QoS, Quality of Service, is used for prioritizing the packets based on urgency
which is set in part of the packet. It’s very important in the VoIP communication
environment since data can always be re-sent so it’s not urgency at all.

Figure 27 Page of QoS

Please note that selecting QoS can not guarantee the voice transmitting will be
expedited. It just can ensure the packets through ZT-1000 will be filtered based on
QoS type. If your router or other network devices do not support QoS, your voice
packet will be queued somewhere between caller and callee.

Figure 28 QoS Type Selection

3.4.6 Others

“SIP Expiry Time” is about how long the registration is valid, and when the time
interval is expired, ZT-1000 will resubmit the registration again to the SIP server
then.

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 Figure 29 Page of SIP Settings - Others

3.5 NAT Traversal

For sharing the Internet access, most of the routers provide a function called NAT,
Network Address Translation. It helps many computers / network devices can use
just one public IP address. However, it’s not friendly to VoIP devices. NAT
Traversal is for helping VoIP devices to get through the NAT successfully in order
to complete the voice communication.

3.5.1 STUN Setting

If your ZT-1000 is behind a router, we suggest you to enable the STUN setting and
then fill the correct information into required fields accordingly. It will help your
ZT-1000 working properly behind the NAT.

Figure 30 Page of NAT Traversal

3.6 Others

3.6.1 Auto Configuration

The function is only applicable when your ZT-1000 needs to get the configuration
file that is provided by specific service providers, like RingStar or UTStarcom. If it
is the case, then you don’t need to configure the SIP related settings. For more
information, please contact with your service provider.

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 Figure 31 Page of Auto Configuration

The auto configuration field description:

Field Description
Service Provider Select the service provider.
RingStar or UTStarcom.
Auto Select how to get the configuration file.
Configuration TFTP first: Get configuration file from TFTP server. If
failed, get configuration file from HTTP server later.
HTTP first: Get configuration file from HTTP server. If
failed, get configuration file from TFTP server later.
Disable: Do not get configuration file by provision.
Get Provision Select how to get the provision server’s information.
Server Auto: Get provision server’s information from DHCP
option. Make sure your DHCP server support DHCP
option and your network type is DHCP client.
Manual: Get provision server’s information from manual
setting.
TFTP Server The IP address or URL with file path of the TFTP
Address provision server.
TFTP Server Port The port of the TFTP provision server.
HTTP Server The IP address or URL with file path of the HTTP provision
Address server.
HTTP Server Port The port of the HTTP provision server.
Upgrade Select how to get the upgrade information.
Firmware File Auto: Get the upgrade information from configuration
file.
Manual: Get the upgrade information from manual
settings. For more security it will be disabled
automatically after manual upgrade once.
Disable: Disable upgrade by provision.
Upgrade TFTP The IP address or URL with file path of the TFTP
Address upgrade server.
Upgrade TFTP The port of the TFTP upgrade server.
Port

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 File Name The file name of firmware.
Provision Expiry The system will get configuration file again after expiry
Time time.

3.6.2 Country Setting

The country code setting is used for selecting the type of your phone set. Most of
the cases, you just need to choose the country you live. If you ever lived in foreign
country, and you carry specific phone set, such as cordless phone you bought in
Germany and found it got difficulties with your ZT-1000, and then you can try
choosing Germany as the setting of country selection.

Figure 32 Page of Country Setting

3.6.3 Call History

This page will show you the incoming and outgoing phone calls. The Ans field in
the Incoming Calls table indicates whether you received such phone call or not.
The Ans field in the Outgoing Calls table indicates the similar thing, if the call was
answered by the counter party.

Figure 33 Page of Call History

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 3.6.4 Dial Plan
Dial Plan is used for setting the specific kind of dialing pattern. Any number you dial not
matching the pattern won’t work. You can place calls in a short way as the rules define.

Figure 34 Page of Dial Plan

The rules can be only one rule or a set of rules “rule 1|rule 2|rule 3”. Maximum length is
60 characters. Each rule should be started with <prefix number> then followed a
regular rule.

Fields definition:
Field Description
<01155> prefix number 01155
[2-6] A number between 2~6
[3478] A number 3 or 4 or 7 or 8
X A number 0~9
0,1,2,3,4,5,6,7,8,9 A number 0,1,2,3,4,5,6,7,8,9

Example : <01155>[1-9]12xx
<01155> means prefix number 01155.
[1-9]12xx is a regular rule with total 5 digits, first number 1~9, second number 1, and
third number 2.
When you dialed 5 digits that accord with the regular rule (ex : 11267),it will add the
prefix “01155” automatically and dial out.(Really dialed out:0115511267)

3.7 System Administration

3.7.1 System Authority

In this page, you can change the default username and password accordingly.
Please note that the password is case sensitive.

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 Figure 35 Page of System Authority

3.7.2 Firmware Upgrade

We do not recommend you to upgrade the firmware of your ZT-1000, unless you
encountered some problems which can be solved by new firmware.

For upgrading the firmware, simply click the “Browse...” button to pick the firmware
image you downloaded from the source. After doing that, just click the “Upgrade”
button.

NOTE: Make sure your power supply is stable during the firmware upgrade, otherwise your
ZT-1000 might be damaged for good.

Figure 36 Page of Firmware Update

3.7.3 Revert to Default

If you have changed something and found ZT-1000 can not work properly any
more. You can use this function to revert all the settings back to factory default

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 values. However, if you do so, all the other necessary information should be
re-entered again.

Figure 37 Page of Revert to Default

3.8 Apply & Reboot

By selecting the “Apply & Reboot,” all the setting changes you made will be
affected. Some of the setting changes will cause system reboot. In such cases,
please wait around one minute, and your ZT-1000 will back to work then.

3.9 Logout

You could logout the webpage by selecting the “Logout”. There could be only one
user login to the webpage at the same time. Any login user will be logout
automatically if this user idles for 3 minutes.

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 Part III Illustrations

Topics in Part III:

z Ch. 4 Troubleshooting

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 Ch. 4 FAQ & Troubleshooting
This chapter highlights some potential problems and the corresponding reactions.

4.1 How can I make a phone call?

Make sure you have done all the settings of your ZT-1000, and the LED is on,
please refer to Section 2.3.

Just pick up the phone and dial the number and ending with the pound key, then
you can talk the counter part you’re party once he/she answers the call.

4.2 How can I pick up another call while I’m on the phone?

If you can use this function, it means the Call Waiting function has been turned on.
When there’s a new incoming call while you are on the phone with the other
person, you can press the Flash button to switch to the new call.

Once you’ve switched the phone conversation to the 2nd person, the first person
you talked with will be put on-hold. You can press the Flash button again to switch
back to the first one.

4.3 How can I make a 3-way conference call?

Making 3-way conference call is pretty similar to making phone call; just follow the
steps below, and your 3-way conference will be there for you.

1. Make a phone call to the first phone number.

2. After the call is established, press the Flash button. You will hear the Dial tone,
and then make the other phone call to the second phone number.
3. When the second call is established, press the Flash button again. Then, all
the 3 parties can enjoy the conference call.

Please note that ZT-1000 only supports 3-way conference function for one phone
at any time. If phone 1 is in 3-way conference, phone 2 won’t be allowed to use
this function until the 3-way conference in phone 1 is finished. During the 3-way
conference establishment, the call to the 2nd callee will use G.711u or G.711A
only.

4.4 How can I transfer a call to others?

For using “Call Transfer” function, you just need to dial "＊"+ "1" to hold the current
call, then the extension number where you like the call to be transferred.

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 4.5 Why can’t I login to the RingStar ZT-1000?

Make sure you set the correct IP address of ZT-1000. For getting the IP address of
your ZT-1000, you can use the IVR instruction, “#120”. If ZT-1000 still won’t
answer you the IP address, please refer to Chapter 3 to do proper settings.

If you’ve confirmed the IP address of ZT-1000 is correct but you still can not
access the webpage of ZT-1000, you probably did not enter the correct port
number, 9999. Check it again, and enter http://<ip address of ZT-1000>:9999
should solve the problem. Otherwise, please contact your vender.

4.6 It seems something wrong about the LED?

Please refer to Chapter 2 for specific LED definition first. If the LED behavior is
different from what described in Chapter 2, and ZT-1000 is malfunction, please
contact you vender.

4.7 Why can’t I access the Internet?

Please check which mode, “Bridge Mode” or “NAT Mode” your ZT-1000 is set, and
check if the Ethernet ports are used correctly. In addition, please refer to Section
3.3 for better understanding of ZT-1000.

4.8 Why can’t I make VoIP phone calls to others?

Please check the “SIP Settings” which is described is Section 3.4, and compared if
all the information your ITSP gave you has been entered into ZT-1000 correctly.

4.9 ZT-1000 does not respond correctly to key press sometimes.

If the behavior of ZT-1000 is not so properly especially you need to dial the
number after the phone connection has established, such as dialing the extension
numbers, the DTMF setting is probably wrong. Please refer to Section 3.4 about
“SIP Settings,” and check the DTMF settings. The setting should be absolutely the
same as what ITSP told you; otherwise, you ZT-1000 might have some troubles
while you need to dial again during the phone calls.

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 Appendix

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 App. A List of Figures
FIGURE 1 FRONT VIEW OF RINGSTAR DUAL-FXS ATA, ZT-1000 ...............................................................6
FIGURE 2 REAR VIEW OF ZT-1000 ...................................................................................................................7
FIGURE 3 WEB CONFIGURATION ENVIRONMENT OF RINGSTAR ZT-1000....................................................8
FIGURE 4 RECOMMENDED NETWORK TOPOLOGY FOR CONFIGURING ZT-1000 .........................................10
FIGURE 5 LOGIN PAGE OF ZT-1000 FOR CONFIGURATION ...........................................................................12
FIGURE 6 MAIN PAGE OF ZT-1000 WEB CONFIGURATION ENVIRONMENT..................................................13
FIGURE 7 PAGE OF PHONE BOOK ...................................................................................................................13
FIGURE 8 PAGE OF PHONE BOOK (CONTINUED) ...........................................................................................14
FIGURE 9 PAGE OF SPEED DIAL ......................................................................................................................15
FIGURE 10 PAGE OF CALL FORWARD ..............................................................................................................16
FIGURE 11 PAGE OF VOLUME ADJUSTMENT ...................................................................................................16
FIGURE 12 PAGE OF CALL BLOCKS ..................................................................................................................17
FIGURE 13 PAGE OF CALLER ID .......................................................................................................................18
FIGURE 14 PAGE OF CALLER ID (CONTINUED) ...............................................................................................18
FIGURE 15 PAGE OF AUTO DIAL.......................................................................................................................18
FIGURE 16 PAGE OF FLASH TIME .....................................................................................................................19
FIGURE 17 PAGE OF CALL WAITING ................................................................................................................19
FIGURE 18 PAGE OF NETWORK STATUS ..........................................................................................................20
FIGURE 19 PAGE OF NETWORK MODE - BRIDGE MODE .................................................................................21
FIGURE 20 PAGE OF NETWORK MODE – NAT MODE .....................................................................................22
FIGURE 21 PAGE OF SNTP SETTING ................................................................................................................23
FIGURE 22 PAGE OF ICMP SETTING ................................................................................................................23
FIGURE 23 PAGE OF SERVICE DOMAIN ............................................................................................................24
FIGURE 24 PAGE OF PORT SETTINGS ................................................................................................................24
FIGURE 25 PAGE OF CODEC SELECTION ..........................................................................................................25
FIGURE 26 PAGE OF DTMF ..............................................................................................................................25
FIGURE 27 PAGE OF QOS...................................................................................................................................26
FIGURE 28 QOS TYPE SELECTION ....................................................................................................................26
FIGURE 29 PAGE OF SIP SETTINGS - OTHERS ..................................................................................................27
FIGURE 30 PAGE OF NAT TRAVERSAL ............................................................................................................27
FIGURE 31 PAGE OF AUTO CONFIGURATION ...................................................................................................28
FIGURE 32 PAGE OF COUNTRY SETTING ..........................................................................................................29
FIGURE 33 PAGE OF CALL HISTORY.................................................................................................................29
FIGURE 34 PAGE OF DIAL PLAN .......................................................................................................................30
FIGURE 35 PAGE OF SYSTEM AUTHORITY .......................................................................................................31
FIGURE 36 PAGE OF FIRMWARE UPDATE.........................................................................................................31
FIGURE 37 PAGE OF REVERT TO DEFAULT ......................................................................................................32

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