ArtDio IPS 1000 Network Card User Manual

Voice Internet Phone Gateway
IPS 1000 Series
User Manual
Version:3.1 Update:2004/5/7
ARTDio Company Inc.
IPS 1000 Series VoIP Gateway
Contents
1.
Safety Instructions ............................................................................................................................4
2.
Preface..............................................................................................................................................4
2.1.
What is SIP........................................................................................................................4
2.1.1.
Components of SIP ...................................................................................................5
3.
Package Contents.............................................................................................................................7
4.
Panel Descriptions ............................................................................................................................7
4.1.
Front Panel........................................................................................................................7
4.2.
Rear Panel ........................................................................................................................8
5.
LED Indicators ................................................................................................................................10
6.
Connectors...................................................................................................................................... 11
7.
IDC Connectors (Only for 1008/1016) ............................................................................................ 11
8.
Information required before Installation...........................................................................................12
9.
8.1.
IP Address .......................................................................................................................12
8.2.
SIP Information................................................................................................................13
8.3.
Prepare a password for Web Management.....................................................................13
Installation and Configuration .........................................................................................................14
9.1.
Confirming the Region ID ................................................................................................14
9.1.1.
Phone Setting ..........................................................................................................14
9.1.2.
System console settings (Only 3704/3708/3716) ....................................................15
9.2.
IP Address Settings .........................................................................................................15
9.2.1.
Static IP Mode .........................................................................................................16
9.2.2.
DHCP Mode ............................................................................................................17
9.2.3.
PPPoE Mode ...........................................................................................................17
10.
SIP Configuration ....................................................................................................................23
10.1.
Channels and SIP entity..................................................................................................24
10.2.
SIP Proxy and Register Parameters ...............................................................................25
10.3.
SIP Entity.........................................................................................................................25
10.4.
SIP Outbound Authentication ..........................................................................................26
10.5.
Configure STUN ..............................................................................................................27
10.6.
Check SIP entity Status ...................................................................................................28
10.7.
Phone Book.....................................................................................................................28
10.8.
Make SIP Calls................................................................................................................29
10.9.
Contact Address ..............................................................................................................30
11.
Other SIP Parameters .............................................................................................................31
11.1.
Dialing Plan .....................................................................................................................31
11.2.
Call Forward ....................................................................................................................32
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IPS 1000 Series VoIP Gateway
11.3.
Inbound Authentication....................................................................................................32
11.4.
FAX..................................................................................................................................33
11.4.1.
The devices at two sides are all IPS 1000 series gateway......................................33
11.4.2.
The devices at two sides are IPS 1000 and the other brands .................................33
12.
WEB MANAGEMENT INTERFACE ........................................................................................34
12.1.
BASIC / GENERAL .........................................................................................................35
12.2.
IP SETTING ....................................................................................................................37
12.3.
ADVANCED / GENERAL ................................................................................................39
12.4.
SIP COMMON.................................................................................................................41
12.5.
SIP OUTBOUND AUTHENTICATION .............................................................................44
12.6.
SIP INBOUND ANTHENTICATION .................................................................................45
12.7.
Dialing Plan .....................................................................................................................46
12.8.
STUN...............................................................................................................................48
12.9.
CHANNEL .......................................................................................................................49
12.10.
PHONE BOOK ................................................................................................................51
13.
Use Private IP (Behind NAT)...................................................................................................52
14.
Appendix .................................................................................................................................53
15.
14.1.
Appendix A: Phone-Set Command..................................................................................53
14.2.
Appendix B: Console Command .....................................................................................55
14.3.
Specifications ..................................................................................................................56
14.4.
Mapping table of characters used in PPPoE...................................................................57
14.5.
Region ID ........................................................................................................................58
Contact Information.................................................................................................................59
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IPS 1000 Series VoIP Gateway
1.
Safety Instructions
WARNING
1. Do not attempt to service the product yourself. Any servicing of this product should be referred to
qualified service personal.
2. To avoid electric shock, do not put your finger, pin, wire, or any other metal objects into vents and
gaps.
3. To avoid accidental fire or electric shock, do not twist power cord or place it under heavy objects.
4. The product should be connected to a power supply of the type described in the operating
instructions or as marked on the product.
5. To avoid hazard to children, dispose of the product’s plastic packaging carefully.
6. The phone line should always be connected to the LINE connector. It should not be connected to the
PHONE connector as it may cause damage to the product.
7. Please read all the instructions before using this product.
2.
Preface
The IPS 1000 unit is a personal SIP VoIP gateway developed using the latest in VoIP technology. It is
also very simple to install and easy to operate.
2.1.
What is SIP
Session Initiation Protocol (SIP) is the Internet Engineering Task Force's (IETF's) standard for
multimedia conferencing over IP. SIP is an ASCII-based, application-layer control protocol (defined in
RFC 2543& RFC 3621) that can be used to establish, maintain, and terminate calls between two or more
end points. Like other VoIP protocols, SIP is designed to address the functions of signaling and session
management within a packet telephony network. Signaling allows call information to be carried across
network boundaries. Session management provides the ability to control the attributes of an end-to-end
call.
SIP provides the following capabilities:
Determine the location of the target end point—Supports address resolution, name mapping, and call
redirection.
Determine the media capabilities of the target end point—By using Session Description Protocol (SDP),
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IPS 1000 Series VoIP Gateway
SIP determines the highest level of common services between the end points. Conferences are
established using only the media capabilities that can be supported by all end points.
Determine the availability of the target end point—If a call cannot be completed because the target end
point is unavailable, SIP determines whether the called party is already on the phone or did not answer
in the allotted number of rings. It then returns a message indicating why the target end point is
unavailable.
Establish a session between the originating and target end point—If the call can be completed, SIP
establishes a session between the end points. SIP also supports mid-call changes, such as the addition
of another end point to the conference or the changing of a media characteristic or Codec.
Handle the transfer and termination of calls—SIP supports the transfer of calls from one end point to
another. During a call transfer, SIP simply establishes a session between the transferee and a new end
point (specified by the transferring party) and terminates the session between the transferee and the
transferring party. At the end of a call, SIP terminates the sessions between all parties.
2.1.1.
Components of SIP
SIP is a peer-to-peer protocol. The peers in a session are called User Agents (UAs). A user agent can
function in one of the following roles:
User agent client (UAC)—A client application that initiates the SIP request.
User agent server (UAS)—A server application that contacts the user when a SIP request is received
and that returns a response on behalf of the user.
Typically, a SIP end point is capable of functioning as both a UAC and a UAS, but functions only as one
or the other per transaction. Whether the endpoint functions as a UAC or a UAS depends on the UA that
initiated the request.
From an architecture standpoint, the physical components of a SIP network can be grouped into two
categories: clients and servers.
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IPS 1000 Series VoIP Gateway
Architecture
SIP Clients
SIP clients include the following:
Phones—Can act as either a UAS or UAC. Soft phones (PCs that have phone capabilities installed) and
Cisco SIP IP phones can initiate SIP requests and respond to requests.
Gateways—Provide call control. Gateways provide much functionality. The most common one is a
translation function between SIP conferencing endpoints and other terminal types. This function includes
translation between transmission formats and between communications procedures. In addition, the
gateway also translates between audio and video Codec and performs call setup and clearing on both
the LAN side and the switched-circuit network side.
SIP Servers
SIP servers include the following:
Proxy server—The proxy server is an intermediate device that receives SIP requests from a client and
then forwards the requests on the client's behalf. Basically, proxy servers receive SIP messages and
forward them to the next SIP server in the network. Proxy servers can provide functions such as
authentication, authorization, network access control, routing, reliable request retransmission, and
security.
Redirect server—Provides the client with information about the next hop or hops that a message should
take, then the client contacts the next hop server or UAS directly.
Registrar
server—Processes requests from UACs for registration of their current location. Registrar
servers are often co-located with a redirect or proxy server.
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IPS 1000 Series VoIP Gateway
3.
Package Contents
The IPS 1000 Gateway
x
1
Power Core
X
1
Accessories for fixing support
X
1
System CD-ROM
X
1
5 IDC Connector
X
4
RJ-45 Ethernet Cable
X
1
RJ-11 Telephone Cable
X
1
(For 1008/1016)
(For 1008/1016)
Rubber footer
4.
4.1.
Panel Descriptions
Front Panel
REGISTERED STUN
IPS 1016 Front Panel (16 ports)
REGISTERED
STUN
IPS 1008 Front Panel (8 ports)
IPS 1004 Front Panel (4 ports)
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IPS 1000 Series VoIP Gateway
IPS 1002 Front Panel (2 ports)
4.2.
Rear Panel
There is a button on the rear panel of gateway for special maintenance. Please don’t touch this button
under normal operation.
IPS 1016 Rear Panel (16 ports)
IPS 1008 Rear Panel (8 ports)
IPS 1202 Rear Panel (4 ports)
IPS 1004 Rear Panel (4 ports)
IPS 1400 Rear Panel (4 ports)
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IPS 1000 Series VoIP Gateway
IPS 1103 Rear Panel (4 ports)
IPS 1101 Rear Panel (2 ports)
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IPS 1000 Series VoIP Gateway
5.
LED Indicators
LED
Label
Description
10/100
LNK/ACT
On
Link up
Off
Link down
Flash
Sending/Receiving
Ethernet
data packets
100Mbps
LOOP/RING
FXS
FXO
Device
Alarm
On (LNK is on)
100Mbps
Off (LNK is on)
10Mbps
On
Off hook
Off
On hook
Flash
Ringing out
On
Line is active
Off
Line is inactive
Flash
Ringing in
The red light “On” indicates that system has
some problem; please contact your vender.
Power
“On” indicates that the power supply is
working normally.
CPU/ACT
“On” indicates that the CPU is working
normally.
Registered “On” indicates that all SIP entities are
registered successful.
“Off” indicates that all SIP entities are
registered fail.
“Flash” indicates that one of these SIP
entities is registered fail.
STUN
“On” indicates communicate with STUN
Server once.
“Off” indicates never communicate with
STUN Server.
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IPS 1000 Series VoIP Gateway
6.
Connectors
Ports
Label
Description
Voice Ports
FXS
Connects to a telephone set or fax
machine
FXO
Ethernet
Connects to the phone line
LAN/Internet RJ-45 connector
Ports
MDI-X connects to a Modem
PC
RJ-45 connector
MDI connects to a PC
Console Port
Console
RJ-45 connector/RS-232 Interface
(Only 1004/1008/1016)
7.
IDC Connectors (Only for IPS-1000 series 8/16 ports)
IDC connector is used for the voice interface (FXS and FXO) on the frame model. IDC connector can
easily connect PBX line and telephone wire together to the gateway.
No special tools are required;
please follow the instruction to install:
(Remarks: For IDC connector, it’s better to use No. 24 wire, e.g. CAT 5)
Get the material ready
Insert the insulated wires directly into the
block for wire insertion
Push the block down until it is locked to
flush the conductor with the probe
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Push from here
IPS 1000 Series VoIP Gateway
Cut off the conductor outside the edge to avoid from
causing the circuit shortage
8.
Information required before Installation
You need to prepare the following information before installing the gateway.
8.1.
IP Address
The gateway requires an IP address for operation. Before installation you need to know how to obtain an
IP address from your local ISP. Static IP, DHCP or PPPoE can be used. The following table helps you to
decide what information you need. If your ISP offers static IP, you may need to obtain an IP from MIS
personnel in order to prevent an IP conflict. Otherwise DHCP (most cable broadband providers offer this)
and PPPoE (most ADSL broadband providers offer this) will work fine.
IP Environment
Static IP
Requiring information
Public IP
IP Address
Address
Subnet Mask
Default Gateway
It is strongly suggested that you obtain an
IP address from MIS personnel in order to
prevent an IP conflict.
Private IP
IP Address
Address
Subnet Mask
Default Gateway
It is strongly suggested that you obtain an
IP address from MIS personnel in order to
prevent IP conflicts.
Your private IP requires an IP Sharing
device and you must configure the IP
Sharing device to treat the IPS unit and the
IP that it is using as a virtual server.
Dynamic IP address (DHCP)
DHCP mode
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IPS 1000 Series VoIP Gateway
PPPoE
Account Number
Password
Your ISP normally provides this information.
If you don’t have this information please
contact your ISP.
8.2.
SIP Information
Before configuring SIP, the IPS 1000 requires SIP information for operation. The following table helps
you to decide what information you need.
Items
Description
1. SIP Proxy
If you want to make SIP calls through SIP proxy
server, you will need to know the IP address or
domain name of SIP proxy server. The proxy
server is an intermediate device that receives
SIP requests from a client and then forwards
the requests on the client's behalf. If you don’t
know which SIP proxy for setting, contact your
SIP service provider.
2. Public Address (SIP Account) The public address is like phone number, you
Example: sip@artdioinc.com
can get the account from your SIP service
provider.
3. Outbound Authentication
You will need the information when the SIP
proxy server requires authentication. You can
get this authentication information from SIP
service provider when you apply for the service.
8.3.
Prepare a password for Web Management
You will need to prepare a password for Web based Management. It can be a digit and/or letter
combination ranging from 1 to 6 digits (E.g. 123). For security reason, password must be set to enter the
Web Management page.
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IPS 1000 Series VoIP Gateway
9.
Installation and Configuration
After preparing the information you need as specified in section 5, follow the following steps to do the
basic configuration. You can use either a telephone or a system console to perform basic configurations.
It is simple to connect a telephone set to FXS port and configures the system. If you want to use system
console to configure the system (Only 1004/1008/1016 support), you have to configure your VT100
terminal to match the settings of the IPS unit’s console port. The console port’s terminal connection is
set to 9600 baud, 8 data bits, 1 stop bit and no parity. Turn on the IPS
unit’s power and wait for the
terminal to display “Press Enter…” follow the directions to begin.
Here are several procedures to do:
1. Confirming the Region ID.
2. Configure IP address of gateway.
3. Enter into the WEB page.
4. Plan and configure the channels into SIP entity.
5. Configure SIP proxy and register information.
6. Configure SIP entity information.
7. Configure Outbound Authentication (If needs).
8. Configure STUN (If your gateway is behind NAT).
9. Check the SIP entity if is registered successful.
10. Configure Phone book (If needs)
11. Make a SIP call.
9.1.
Confirming the Region ID
Skip this step if you are installing your IPS unit in the default region. The default Region ID is printed on
the label located outside the box. If you are installing your IPS unit at any region other then the region ID
specified on the label, you will then need to configure the IPS to the correct Region ID.
9.1.1.
Phone Setting
1. Connect the power.
2. Connect the phone cable to the “Phone” socket on the rear panel as pictured above.
3. When the CPU/ACT LED is on, pick up the handset and listen for the dialing tone.
4. Dial “##0000” and listen for 3 short beep.
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IPS 1000 Series VoIP Gateway
5. Dial “9507#”;Assuming you are modifying for China (The last 2 digits are the regional ID)
6. Dial “971#”;Sets the new regional ID.
7. Hang up the phone. The device will be updated with the new region setting after it restarts (restart
time is about 10 seconds)
9.1.2.
System console settings (Only for 4/8/16 ports)
SIP-RG>enable
SIP-RG #configure
Enter configuration commands, one per line.
End with CNTL/Z
SIP-RG(config)#regional_id 07
SIP-RG(config)#exit
SIP-RG#delete nvram
This command resets the system with factory defaults.
All system parameters will revert to their default factory settings.
All static
and dynamic addresses will be removed.
Reset system with factory defaults, [Y]es or [N]o? Yes
Attention:
Before Changing the Region ID, the system has to be reset to the default value. Therefore this step
should be done first.
The following instruction may keep the IP address unchanged after reset:
“delete nvram keep_ip”
9.2.
IP Address Settings
We recommend using a traditional phone to configure the unit’s parameters, as this is the easiest way.
The following two sections contain the procedures used to configure the IPS unit according to how you
obtain your IP address (Static IP; DHCP or PPPoE).
Attention:
Every time you set a parameter item and press the “#” key to complete it, a successful setting will be
confirmed by three equal tones in succession. If your setting is unsuccessful you will be prompted with
one long tone.
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IPS 1000 Series VoIP Gateway
9.2.1.
Static IP Mode
The following table shows an example.
IP Address
210.67.96.121
Subnet Mask
255.255.255.248
Default Gateway
210.67.96.120
Web Management Password
123
Using the information contained in the example above. The procedure is as follows:
1. Connect the IPS unit to a suitable Power source.
2. Connect a traditional phone set to the “FXS” connector located on the rear panel.
3. When the CPU/ACT light is on, pick up the phone to hear the dialing tone.
4. ##0000
; you should hear three short tones.
5. 010#
; the digit “0” is used to enable “manual” IP mode.
6. 02210*67*96*121#
; IP address.
7. 03255*255*255*248#
; Subnet Mask.
8. 04210*67*96*120#
; Default Gateway.
9. 15123#
; “123” is the web management password.
10. 981#
; Warm-restarts.
11. Hang up the phone. The system should now restart.
You can also use console to configure IP address. But phone number can’t be configured by
console.(Only 1004/1008/1016)
SIP-RG>enable
SIP-RG#configure
Enter configuration commands, one per line.
End with CNTL/Z
SIP-RG(config)#ip state user
SIP-RG(config)#ip address 210.67.96.121 255.255.255.248
System need to restart
SIP-RG(config)#ip default-gateway 210.67.96.120
SIP-RG(config)#exit
SIP-RG#restart
This command resets the system.
System will restart operation code agent.
Reset system, [Y]es or [N]o? Yes
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IPS 1000 Series VoIP Gateway
9.2.2.
DHCP Mode
1. Connect the IPS unit to a suitable Power source.
2. Connect a traditional phone set to the “FXS” connector located on the rear panel.
3. When the CPU/ACT light is on, pick up the phone to hear the dialing tone.
4. ##0000
; you should hear three short tones.
5. 011#
; the digit “1” is used to enable “DHCP” IP mode.
6. 15123#
; “123” is the web management password.
7. 981#
; Warm-restarts.
8. Hang up the phone. The system should now restart.
You can also use console to configure IP address. But phone number can’t be configured by
console.(Only for 4 ports gateway)
SIP-RG>enable
SIP-RG#configure
Enter configuration commands, one per line.
End with CNTL/Z
SIP-RG(config)#ip state dhcp
SIP-RG(config)#exit
SIP-RG#restart
This command resets the system.
System will restart operation code agent.
Reset system, [Y]es or [N]o? Yes
9.2.3.
PPPoE Mode
If your network environment is using PPPoE, you need to prepare the information as specified in section
8. Information required before Installation.
The following table shows an example.
PPPoE Account
83721@hinet.net
PPPoE Password
123ab
Web management password
123
There are three ways to configure user name and password of PPPoE
1. Use phone set to configure:
You can configure the user name and password by using phone set. The command ‘09’ is used for
username and ‘10’ is for password of PPPoE. Since the user name and password use characters and
17
IPS 1000 Series VoIP Gateway
digits are accepted by phoneset only, you need a mapping between characters and digits. You can find
them at section 14.4 Mapping table of characters used in PPPoE.
Example user name:83721@hinet.net,Password:123ab,The procedure is below
1. Connect the phone to IPS
2. When CPU/ACT is light, pick up the phone
and press
;You will hear 3 short tones.
3. ##0000
4. 0938333732314068696*465742*46*46574# ;Set user name:83721@hinet.net
5. 103132336162#
;Set password is 123ab
6. 981#
;Save and restart.
2. Use Console to configure (Only for 4/8/16 ports Gateway)
SIP-RG>enable
SIP-RG#configure
Enter configuration commands, one per line.
End with CNTL/Z
SIP-RG(config)#pppoe username 83721@hinet.net
SIP-RG(config)#pppoe password 123ab
SIP-RG(config)#exit
SIP-RG#restart
This command resets the system.
System will restart operation code agent.
Reset system, [Y]es or [N]o? Yes
3. Use WEB Interface to configure:
You can configure the user name and password by using WEB interface. Follow the steps to finish
configuration.
Step 1: Using a traditional phone set to configure the web management password and phone number
You will need to use a web browser to perform the PPPoE settings through the IPS
unit’s web based
management interface. To enter the web based management interface you must have a previously
configured password. Follow the next procedure to setup your password and phone number.
1. Connect the IPS unit to a suitable Power source.
2. Connect a traditional phone set to the “Phone” connector located on the rear panel.
3. When the CPU/ACT light is on, pick up the phone. You should hear the dialing tone.
4. ##0000
; you should hear three short tones.
5. 15123
; “123” is the web management password.
6. 010#
; “0” is to enable “manual” IP mode.
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IPS 1000 Series VoIP Gateway
7. 02192*168*0*2#
; IP address.
8. 03255*255*255*0#
; Subnet Mask .
9. 981#
; Used to restart the IPS unit.
10. Hang up the phone to complete the configuration.
Step 2:Configure IP address of PC
Use the provided Ethernet cable to connect your PC to the port labeled “PC”, located on the rear panel
of the IPS unit.
Because the IPS-1000 series unit’s default IP setting is 192.168.0.2, you must configure your PC to the
same subnet. “192.168.0.x” for example. The following example uses 192.168.0.5 for the IP address
and 255.255.255.0 for the subnet mask.
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IPS 1000 Series VoIP Gateway
After you have completed the PC’s IP address setting, you will be required to restart the PC in order for
the new settings to take effect.
Step 3: Using the browser to configure the PPPoE Parameters of the gateway.
On the PC that is connected to the IPS unit, enter the IPS unit’s IP address (Default 192.168.0.2) and
press enter. The IPS will then prompt you with a dialogue box requesting that you enter a password. Use
“WEB” (all capitals), for the User field and “123” for the password field that you have previously
configured. Click the OK button; you should now have access to the IPS unit’s web based management
interface page.
The
MOSA
unit’s IP
“WEB”
should be all
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IPS 1000 Series VoIP Gateway
Upon entering the web based configuration interface.
Click on “IP SETTING” at the top of the page and you will see the page as shown in the following image.
Select PPPoE from the “IP State” pull down menu.
Fill in the “Account”, “Password”, and “Confirm Password” under the PPPoE Settings. You can obtain
this information from your ISP.
Click on the Apply button.
Click the “BASIC” button at the top to go to the BASIC page and select “Warm Start” to restart the
gateway. You can also perform a warm start using the phone by picking up the handset and dialing
“##0000” then “981#”.
After restarting, the gateway will use PPPoE to obtain it’s IP address.
1
Click “IP setting”
4
to open this
2
display
Click the “Apply”
3
button to apply any
changes.
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IPS 1000 Series VoIP Gateway
6
Click the “Apply”
button to apply
any changes.
5
At this stage, your IPS should be able to use PPPoE to access the Internet. However, if you configured a
wrong account number or password, your IPS cannot access the Internet. You are not able to use PC to
access IPS by using the IP address of 192.168.0.2 because IPS has been set in PPPoE mode. You
have to use phone set to configure IPS back to fix IP mode (##0000 010#) and use PC browser to
configure correct parameters.
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IPS 1000 Series VoIP Gateway
10.
SIP Configuration
IPS 1000 not only can make regular PSTN calls, it also can communicate with IP Phones or Soft-Phones
by using SIP protocol. Previous paragraphs have described the way to make regular IP calls. This
section shows you what parameters you need to configure for SIP calls and how to make the SIP calls.
SoftPhone ( Notebook /PC )
IPS 1000
IPS 1000 (SIP)
IP
Cisco IP Phone
Notice: These configurations on WEB page, after select or input value in the field, please press “Apply”
button to save and confirm the setting. Some parameters need “Warm-restart”, please process the
restart action, thanks.
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IPS 1000 Series VoIP Gateway
10.1.
Channels and SIP entity
Select the channel and join a SIP entity.
Figure:
-
IPS-1008
SIP Entity 2
SIP Entity 3
SIP Entity 4
FXS
SIP Entity 1
SIP Entity 5
Configuration:
WEB page: CHANNEL\
Notice: Each channel must belong to a SIP entity.
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IPS 1000 Series VoIP Gateway
10.2.
SIP Proxy and Register Parameters
You need to configure IP address or Domain name of Registrar and Outbound Proxy server, please
check the information is right.
SIP service provider will give you an IP address or Domain name of Registrar and Outbound proxy when
you apply for the service.
Configuration
WEB Page: ADVANCED\SIP COMMOM
Notice: The Registrar Server is only for SIP entity registering. If the SIP entity register is fail, please
check the item. SIP calls are all through Outbound Proxy Server, if the parameter is not configured, the
SIP call will fail. So the two parameters must be configured.
10.3.
SIP Entity
SIP service provider will assign one or more SIP accounts for you when you apply for the service. In
standard, the SIP account is called ‘Public Address’, so you need to configure the account information in
‘Public Address’ item. The format is like an E-mail address such as mary@artdioinc.com.
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IPS 1000 Series VoIP Gateway
Configuration
WEB Page: ADVANCED \ SIP COMMON
You can control the SIP entity on WEB page, just select ‘Enable’ or ‘Disable’.
10.4.
SIP Outbound Authentication
You need to configure outbound authentication for each SIP entity if SIP proxy server or other SIP phone
request for authentication. Please check with SIP service provider if you need the setting.
Please select the entity then input information includes realm, username, and password.
Configuration
WEB Page: ADVANCED \ SIP OUTBOUND AUTHENTICATION
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IPS 1000 Series VoIP Gateway
10.5.
Configure STUN
The STUN (Simple Traversal UDP through NAT) server is an implementation of the STUN protocol that
enables STUN functionality in SIP-based systems. The STUN server also includes a client API to enable
STUN functionality in SIP endpoints.
STUN is an application-layer protocol that can determine the public IP and nature of a NAT device that
sits between the STUN client and STUN server.
Notice: If your gateway is behind NAT (Use Private IP), must configure the parameter.
After configuring the parameters of STUN, please act Warm-Restart.
Configuration
WEB Page: ADVANCED\STUN
You can enable and disable the service on WEB page.
The STUN refresh time defines how long the device will send a binding request packet with discard flag
on to STUN server. A binding packet with discard flag off will be sent each time when the number of
binding request packet with discard flag on reach the Rebinding counts. The binding request packet is
used to let the STUN server keep the most fresh client information.
27
IPS 1000 Series VoIP Gateway
10.6.
Check SIP entity Status
You can use the WEB page to check the SIP entity is registered successful or unsuccessful.
WEB Page: ADVANCED\SIP COMMOM
If the status shows “REGISTERED” means successful, otherwise means fail; please notice that.
When you find the registration is fail, first check the “Registrar Setting” configuration is normal, or not
“Enable”.
Then check the “Public Address” and “Outbound Authentication” configuration is in normal status.
If the configurations are all right, please check the situation with your SIP service provider.
10.7.
Phone Book
Since the SIP phone number is not easy for regular phone to dial, IPS 1000 provide a SIP phone book to
let standard phone to make a SIP call easier. The phone book uses index number to map SIP account.
For instance if the phone book is configure as below:
Index
Public Address
Port
Proxy
2231
mary@artdioinc.com
5060
Yes
331
John@artdioinc.com
5060
No
Notice: If your SIP account is number type like 234@artdioinc.com or 456@artdioinc.com, you don’t
need to configure the items.
Configuration
WEB page: PHONEBOOK \
28
IPS 1000 Series VoIP Gateway
10.8.
Make SIP Calls
After you have configured the SIP phone on the SIP phone book, you can easily make SIP calls.
You can select one way to make SIP call following these ways:
Standard Call: Only dial <numbers>+<#>.
1. Compare dialing plan, check the number if it is in setting. Example 050.
2. If the number is in setting, send the call to proxy. If communicate with proxy is fail, then the call will
be sent to PSTN.
3. If the number is not in dialing plan, the call will be sent to PSTN.
Force SIP Call: Dial <#>+ <numbers>+<#>.
1. Compare SIP Phone books; check the number if it is in phone book.
2. If the number is in setting and Proxy selection is set to "No", you will hear a busy tone. If Proxy
selection is set to "Yes", then send the call to proxy.
3. If communicate with proxy is fail, you will hear a busy tone.
4. If the number is not in phone book, you will hear busy tone.
Force PSTN Call: Dial <*>+<numbers>+<#>.
Always go through PSTN
Notice: If you do not want to dial “#” after numbers, please configure the ‘Dial Ending Time’ item. After
the seconds, the call will be sent automatically.
WEB Page: ADVANCED\GENERAL
29
IPS 1000 Series VoIP Gateway
10.9.
Contact Address
The main purpose of Contact Address is making SIP calls without proxy.
The Contact Address can be any numbers or characters such as ‘Mary’ or ‘1003’.
WEB Page: ADVANCED\SIP COMMOM
Making SIP calls without proxy server:
The SIP protocol allows you to make SIP calls directly to the destination number without through the
proxy server. You can simply dial the SIP number and domain name or IP address. The typical example
is: mary@artdioinc.com or 1003@artdioinc.com.
Notice: For this type of SIP calls, the destination device’s IP address is already known and fixed.
30
IPS 1000 Series VoIP Gateway
11.
Other SIP Parameters
11.1.
Dialing Plan
X means all calls will be send to SIP proxy first, if the SIP call is fail, and then sent to PSTN.
If the configuration is only ‘050’ means the numbers like 050xxxxx will send to SIP proxy, if you dial any
other numbers like 100, the number will send to PSTN immediately.
C
O
Dialing Plan:
050 and 070
FXO
Dial 82261234
IPS 1000
The call is sent to
FXS
Dial 050123456 or 070345678
The call will be defined to SIP account
PSTN immediately
and sent to SIP Proxy. If the SIP call is
fail, then send to PSTN.
Configuration
WEB Page: ADVANCED\Dialing Plan
31
IPS 1000 Series VoIP Gateway
11.2.
Call Forward
There are three forward types:
1. All: All incoming call to the SIP entity will be forward.
2. Busy: When the SIP entity is busy, the incoming call will be forward.
3. No Answer: When the SIP entity is no answer and after 30 seconds, the incoming call will be
forward.
Notice: In order to let the caller identify the port has been configured ”forward”; the caller will hear
second dial tone, rather than normal dial tone.
Configuration
WEB page: ADVANCED\SIP COMMOM
Phone Set: Please refer to section Appendix A: Phone-Set Command.
11.3.
Inbound Authentication
You need to configure inbound authentication if you request authentication for other SIP phone to call
you.
Configuration
WEB Page: ADVANCED \ SIP INBOUND AUTHENTICATION
32
IPS 1000 Series VoIP Gateway
11.4.
11.4.1.
FAX
The devices at two sides are all IPS 1000 series gateway
Use the FAX protocol that is the proprietary protocol of IPS (supporting T.38). Setup method is listed
below:
1. Web Folder: “Connect Device” in “Channel” folder. Select “FAX” and then click “Apply” button
2. Web folder: “IPS
Protocol” in “Basic” folder
Signaling Port: input “2000”
Support T.38: select “Yes”
Click “Apply” button
3. Warm-Restart the system
11.4.2.
The devices at two sides are IPS 1000 and the other brands
Use the FAX protocol as G.711 (non-supporting T.38). Setup method is listed below:
1. Web folder: “Connect Device” in “Channel” folder. Select “FAX” and then click “Apply” button
2. Setup “Check Protocol”, web folder: ADVANCED\SIP COMMON
Select and mark “PCMU” and “PCMA” Codecs, than click “Apply” button
33
IPS 1000 Series VoIP Gateway
3. Web folder: “IPS Protocol” in “Basic” folder
Signaling Port: input “0”
Support T.38: select “No”
Click “Apply” button
4. Warm-Restart the system
12.
WEB MANAGEMENT INTERFACE
The Tree Architecture of Web Management
HOME
BASIC
GENERAL
IP SETTING
ADVANCED
General
SIP COMMON
SIP OUTBOUND
AUTHENTICATION
SIP INBOUND ATHENTICATION
STUN
Dialing Plan
CHANNEL
PHONE BOOK
ACCESS
CODE
34
IPS 1000 Series VoIP Gateway
12.1.
BASIC / GENERAL
Category
Section
Description
Default Setting
Information
Region ID
Display region ID.(Read only)
0
Software
Display software version.(Read only)
Version
BootRom
Display BootRom Version.(Read only)
Version
Hardware
Display hardware Version.(Read only)
Version
Card Type
Display card type. (Read only)
35
IPS 1000 Series VoIP Gateway
Up-Time
Display the use time since from system
reboot.(Read only)
MAC
Display MAC address.(Read only)
Address
Time
Date
Show the date
Time
Show the time
Date
Manually Input date, only effected in
Configuration
Empty
Manual Mode.
yyyy / mm / dd
Time
Manual input time, only effected in
Empty
Manual Mode of Time Source.
hh : mm : ss
Time Zone
Select local system time zone. Select
correct Time Zone.
Daylight
ON: Enable daylight saving.
OFF
saving
OFF: Disable daylight saving.
IPS
Signaling
UDP port to transfer signal packets. It
protocol
Port
can be setting in the range of 0 to
0
65535. (Must reboot system to apply
changes)(Only support IPS
device)
RTP
Base of UDP port to receive RTP
Base Port
packets. It can be setting in the range of
4000
0 to 65534.( Must be Even, after setting
this item, please reboot system to apply
changes)
Support
Enable/Disable the FAX relay (T.38) of
T.38
IPS
System
Restart
None: Not to restart system.
Restart
Mode
Cold restart: Cold restart.
No
Protocol
Warm restart: Warm restart.
36
None
IPS 1000 Series VoIP Gateway
12.2.
IP SETTING
Category
Section
Description
Default Setting
IP Settings
IP State
The way to obtain IP address:
Manual
Manual: Entered by user
(Static IP)
Auto(DHCP): Assigned by
DHCP server
PPPoE: Assigned by PPPoE of
ISP
37
IPS 1000 Series VoIP Gateway
Current Setting
Display the configured IP
192.168.0.2
address, subnet mask address
255.255.255.0
and default gateway. (Read
192.168.0.1
only)
Change To
Enter the IP address that will
be used after next restart,
Including:
IP Address
Subnet Mask Address
Default Gateway
(This item is used only on
Manual mode of IP Setting.)
PPPoE
Account
Settings
The user’s account of PPPoE
protocol, provided by ISP.
Password
The user’s password of PPPoE
protocol.
Confirm
Confirm the user’s password of
Password
PPPoE protocol.
Service Name
The service name of PPPoE
account, provided by ISP.
(Most ISP doesn’t need this)
DNS Server
Primary Address
The primary address of DNS
168.95.1.1
server. The default setting
would be different according to
the local area. In Taiwan, the
default setting is 168.95.1.1.
Web
Secondary
The secondary address of
Address
DNS server.
User Name
The user’s name of Web
Password
Management Interface.(12
character)
Password
The password of Web
Management Interface.( 6
character)
Password
Enter the password again to
Confirm
confirm it.
38
WEB
IPS 1000 Series VoIP Gateway
12.3.
ADVANCED / GENERAL
Category
Section
Description
Default Setting
Flash Button
Flash Time
System confirmed
200 msec
“Flash” time.
Touch Tone (DTMF)
Duration
The duration to send a
100 msec
DTMF.
Inter-digit
The inter-digit time of
100 msec
sending string of DTMF
digits.
Guard Time
Line
The time defines how
0.8 sec
long the system will not
take incoming call after
call has been
disconnected.
Dial Ending Time
Dial Ending
The time specifies how
4
Time
long to end the dialing
1-10 (seconds)
number if a ‘#’ digit is
missing.
Busy Tone Spec
Frequency
f1, f2
(300 ~ 3000Hz)
39
IPS 1000 Series VoIP Gateway
Cadence
on, off. The on and off
(100 ~ 5000ms)
duration in playing the
tone
Reorder Tone Spec
Frequency
f1, f2
(300 ~ 3000Hz)
Cadence
on, off. The on and off
(100 ~ 5000ms)
duration in playing the
tone
40
IPS 1000 Series VoIP Gateway
12.4.
SIP COMMON
41
IPS 1000 Series VoIP Gateway
Section
Item Field Description
Default
Port and Header
Port
The control port number of SIP protocol. 5060
Header
Select ‘Standard’ or ‘Compact’ to be the Standard
Form
header format of SIP packet. When
Compact is selected, the header will be
shorter and it saves bandwidth.
Outbound Proxy
Domain
Setting
Name
Domain name or IP address of proxy.
Empty
Disable
Port
Control port number of SIP protocol.
5060
Domain
Domain name or IP address of proxy
Empty
Name
that you want to register.
Disable
Out-band DTMF
Control
Enable/Disable
Disable
Codecs Selection
Codec
G.729AB: Mark the selection to Enable Enable
Registrar Setting
Type
G.729AB Codec
G.723.1:
Mark the selection to Enable Enable
G.723.1 Codec
PCMU:
Mark the selection to Enable
Enable
PCMU Codec
PCMA:
Mark the selection to Enable
Enable
PCMA Codec
Codec
You can select the codec priority for
G729-G723-P
Priority
your requirement.
CMU-PCMA
Select an entity
1
SIP Entity
Select: Select Button
Register: Register Button
De-Register: Cancel Register Button
Entity Control
Register Status
Select Enable/Disable
Enable
Register
Show the register status, if it shows
Empty
Status
Registered means successful. (Read
only)
Register: Register Button
De-Register: Cancel Register Button
Public Address
Setting
Address
Enter SIP phone number of the port.
The phone number general assigned by
SIP service provider.
42
Empty
IPS 1000 Series VoIP Gateway
Section
Item Field Description
Contact Address
Name
Setting
Default
Enter Contact Address. You can assign Empty
a name for the port. However, it should
be unique within per channels.
RFC 2833 DTMF
Current
Display current setting of
(Read Only) 01
Setting
Contact Address
2833
Enable/Disable RFC 2833 DTMF.
Never
DTMF
Forward To
2833 In
Display current status of
(Read Only)
Use
DTMF configuration.
Forward
Enter a SIP account (Public Address)
Address
forward. When users dial into the SIP
Empty
Entity, the call will be forwarded to the
number.
Type
N/A: All incoming calls are forward.
N/A
Busy: When the SIP entity is busy, the
calls will be forward.
No Answer: When the SIP entity is no
answer about 30 seconds, the calls will
be forwarded.
SIP Entity
Channel
Show the all channels
Members
Depend on
gateways
Entity
Show ‘+ ‘means the SIP entity is for the Empty
channel.
43
IPS 1000 Series VoIP Gateway
12.5.
SIP OUTBOUND AUTHENTICATION
Section
Item Field Description
SIP Outbound
Maximum Maximum number of entries (Read Only) 50
Authentication
Default
allowed
Entered
Number of entries of
(Read Only) 0
authentication entered.
(Read Only) Empty
Entries
List of entries
List
Entity: Which entity that you select.
Realm: Domain name or IP address.
Username: Username of authentication.
Password: Password of authentication.
44
IPS 1000 Series VoIP Gateway
Section
Item Field Description
Default
Update
Enter the information of outbound
Empty
Entry
authentication
Entity: Select an entity.
Realm: Domain name or IP address.
Username: Enter Username of
authentication.
Password: Enter password of
authentication.
Confirm Password: Enter password again
for confirmation.
Delete
Delete the information of outbound
Entry
authentication
Entity: Select an entity.
Realm: Domain name or IP address.
12.6.
SIP INBOUND ANTHENTICATION
45
Empty
IPS 1000 Series VoIP Gateway
Section
Item Field
Description
Default
SIP Inbound
Realm
Enter domain name or IP address
Empty
Authentication Maximum
Maximum number of
(Read Only) 20
entries allowed
Entered
Number of entries of
(Read Only) 0
authentication entered.
Entries List
Display the entries
(Read Only) Empty
Entity: Which entity that you select.
Username: Username of authentication.
Password: Password of authentication.
Update Entry
Enter entries of authentication
Empty
Entity: Which entity that you select.
Username: Username of authentication.
Password: Password of authentication.
Confirm Password: Enter password
again for confirmation.
Delete Entry
Delete entries of authentication
Entity: Which entity that you want to
delete.
Username: Username of authentication.
12.7.
Dialing Plan
46
Empty
IPS 1000 Series VoIP Gateway
Section
Item Field
Description
Default
DIALING PLAN
Maximum
Maximum number of (Read Only) 100
entries allowed
Entered
Number of entries of (Read Only) 0
authentication
entered.
List
Display the entries
(Read Only) x
The default value “x“means that all
numbers that you dial will first go
through SIP proxy. If the call
communicates with SIP proxy is
fail, it will be transferred to PSTN.
Add Dialing Plan
Enter numbers. Example: 050.
Empty
Delete Entry
Enter numbers for delete.
Empty
47
IPS 1000 Series VoIP Gateway
12.8.
STUN
Section
Item Field
Description
STUN Server
Control
Enable or Disable STUN Server service. Disable
STUN Server
Maximum
Maximum number of
Setting
Default
(Read Only) 5
entries allowed
Entered
Number of entries of
(Read Only) 0
STUN server that have
been entered.
List
Display all of servers that
(Read Only)
have been entered.
Add
Add a stun server
IP Address: Enter IP address or Domain
Name
Port: Enter port number of service.
48
Empty
IPS 1000 Series VoIP Gateway
Section
Item Field
Description
Default
Delete
Delete a stun server
Empty
IP Address: Enter IP address or Domain
Name.
Port: Enter port number of service.
NAT Type
Type
Display NAT type
(Read Only) Unknown
Stun Refresh Time
Interval
It defines how long the device will send 30
a binding request packet with discard
flag on to STUN server.
Mapping List
List
My ip/port: shows the
private IP and port
number.
Global ip/port: Display
public IP and port number.
12.9.
CHANNEL
49
(Read Only) Empty
IPS 1000 Series VoIP Gateway
Category
Section
Description
Default
Setting
Information
Channel
Channel number:
1
Channel
Display port type. (Read only)
Type
Phone: FXS Interface, connect
to telephone set or Fax
machine.
Line: FXO Interface, connect to
phone line.
NA: Not available.
Channel
Enable/Disable all functions of
Control
this port.
Enable
Enable/Disable
Current State
Display the current state of this
port. (Read only)
Enable/ Disable.
Do not
Enable/Disable does not
Disturb
disturb function
Silence
Enable/Disable the function.
Disable
Disable
Suppression
2833 In use
Yes:
(Read only)
No:
Join SIP
Select an Entity for SIP.
1
Connect
Phone: Connect to the FXS
Phone
Device
port is regular phone
Entity
FAX: Connect to the FXS port
is FAX machine
Voice
Input Gain
Adjust Voice input Gain
0
Output Gain
Adjust Voice output Gain
0
50
IPS 1000 Series VoIP Gateway
12.10. PHONE BOOK
Section
Item Field Description
Default
SIP Phone Book Maximum Maximum number of entries (Read Only)
200
allowed
Entered
Number of entries of phone (Read Only)
0
books entered.
Entries
Display phone books
List
Index: Dialing number
(Read Only)
Empty
Public Address: SIP account.
Port: Port number.
Via Proxy: Via proxy or not.
Empty
Update
Enter entries
Entry
Index: Enter dialing number
Public Address: Enter SIP account.
Port: Enter port number
Via Proxy: Select via Proxy or not
Delete
Delete entries
Empty
Entry
Index: Enter the index for delete.
51
IPS 1000 Series VoIP Gateway
13.
Use Private IP (Behind NAT)
Using a Private IP in a NAT Environment
The IPS unit is able to communicate with other IPS units under a NAT environment using Private IP
addresses on the LAN side of your IP Sharing device. However you must configure the IP Sharing
device to treat the IPS unit as a Virtual Server using UDP port 5060, 2000.
You will have to ask MIS personnel to enable the ports listed in the following table.
Packet Modes
Using Ports
SIP Signal Packets
UDP 5060
IPS
Signaling Port
UDP 2000
IPS
RTP Base Port
UDP 4000
FTP software upgrade
TCP 21
Web management
TCP 80
If you want to use private IP behind NAT and Proxy Server is in Internet, you must need to enable STUN
service. If the system is installed in VPN, it is not necessary to Enable Stun.
52
IPS 1000 Series VoIP Gateway
14.
Appendix
14.1.
Appendix A: Phone-Set Command
Pick up the handset and listen for the dialing tone. Dial “##0000 and listen for three consecutive tones
before setting the following parameters. After input the parameters, please dial ‘# to end the
configuration.
Command
Description
Parameters
01
IP State
0 : static; 1: DHCP; 2: PPPoE
02
IP Address
xxx*xxx*xxx*xxx
03
Subnet Mask
xxx*xxx*xxx*xxx
04
Default Gateway
xxx*xxx*xxx*xxx
05
Primary DNS Server
xxx*xxx*xxx*xxx
IP
06
Second DNS Server
xxx*xxx*xxx*xxx
IP
07
Select Signaling Port
0~65535
08
Select RTP Base Port 0~65534 (limit to even port number only)
09
PPPoE username
User name (use the mapping table to map
character into digits)
10
PPPoE password
Password (use the mapping table to map
character into digits)
11
DND
0 : Disable ; 1: Enable
12
SIP Forward State
0 : Disable ; 1: Enable; 2: Busy; 3: No Answer
13
SIP Forward Target
6 Digits
14
Change Service Port
1:FTP; 2:HTTP 3:Telnet (Port: 0-65535)
15
Change WEB
6 digits
Password
16
Change FTP
6 digits
Password
40
Listen for the IP
(ending ”#” is not required)
Address
41
Listen for the Subnet
(ending ”#” is not required)
53
IPS 1000 Series VoIP Gateway
Mask
42
Listen for the Default
(ending ”#” is not required)
Gateway
43
Listen for Current
(ending ”#” is not required)
Signaling Port
44
Listen for Global IP
(ending ”#” is not required)
Address
45
Listen for Global
(ending ”#” is not required)
Signaling Port
46
Listen for WEB, FTP,
1:FTP; 2:HTTP 3:Telnet
Telnet Port
47
Listen for Current
(ending ”#” is not required)
Public Address
96
Region ID
2 digits
97
Reset unit to Factory
1: reset all; 2: keep IP
Default values
98
System Warm Restart 1: do it
54
IPS 1000 Series VoIP Gateway
14.2.
Appendix B: Console Command
User Exec commands
Enable
Turn on privileged commands
Exit
Help
Show
Exit from the EXEC
Description of the interactive help system
Show running system information
show
Dns
ethernet
history
Ip
running-config
version
Show the IP address of domain name server
Fast Ethernet port status and configuration
Display the session command history
Display IP configuration
Show current operating configuration
System hardware and software status
Privileged Mode
Configure
Delete
Disable
Exit
Help
Ping
Probe-hook
Probe-remove
Reload
Restart
Show
Enter configuration mode
Reset configuration
Turn off privileged commands
Exit from the EXEC
Description of the interactive help system
Send echo request to destination
probe busy tone cadence
stop probe busy tone cadence
Halt and perform cold start
Halt and perform warm start
Show running system information
Global Mode
Dbflush
Dns
End
Exit
Help
Ip
Log
No
pppoe
regional_id
service_port
Data Base flush
Set the IP address of domain name server
Exit from configure mode to privileged mode
Exit from configure mode
Description of the interactive help system
Global IP configuration subcommands
Control log output
Negate a command or set its defaults
PPPoE configuration subcommands
Set regional id
Set service port number
55
IPS 1000 Series VoIP Gateway
14.3.
Specifications
Voice Interface
Loop start, 2 wire
FXS interface
Feeding Voltage: 20V
Feeding Current: 30 mA
FXO interface
Connectors
Loop start, 2 wire
RJ-11 Connectors (3702/3704)
IDC Connectors (3708/3716)
Voice compression
G.711/G.723/G.729AB
Silence suppression
VAD, CNG
Echo cancellation
G.165/G.168 16ms
Jitter buffer
Adaptive jitter buffer management
Gain control
In/Out +/-6db
Transport protocols
RTP, RTCP
Call control protocol
Pure SIP
Network Interface
Number of ports
Two Ethernet ports
Interface
10BASE-T/100BASE-TX Auto-negotiation
Connectors
RJ-45 Connectors
General Spec
Dimension
Power
Power consumption
Working environment
IPS
3702: 190mm x 110mm x 25 mm
IPS
3704: 172mm x 177mm x 35 mm
IPS
3708: 440mm x 44mm x 254 mm
IPS
3716: 440mm x 66mm x 254 mm
Voltage: 100-240 VAC, Frequency: 50/60 Hz
IPS
3702: 8 W
IPS
3704: 12W
IPS
3708/3716: 70W
Operating temperature: 0 to 50℃
Storage temperature: -10 to 70℃
EMI
FCC part 15 Class B
PTT
FCC part 68 , NALTE , iDA , JATE
Safety
cUL , CCIB , CB
56
.
CE Mark
IPS 1000 Series VoIP Gateway
14.4.
Mapping table of characters used in PPPoE
Character
Digits to key-in
Character
Digits to key-in
0
30
X
58
1
31
Y
59
2
32
Z
5*0
3
33
a
61
4
34
b
62
5
35
c
63
6
36
d
64
7
37
e
65
8
38
f
66
9
39
g
67
@
40
h
68
A
41
i
69
B
42
j
6*0
C
43
k
6*1
D
44
l
6*2
E
45
m
6*3
F
46
n
6*4
G
47
o
6*5
H
48
p
70
I
49
q
71
J
4*0
r
72
K
4*1
s
73
L
4*2
t
74
M
4*3
u
75
N
4*4
u
76
O
4*5
w
77
P
50
x
78
Q
51
y
79
R
52
z
7*0
S
53
=
3*3
T
54
.
2*4
57
IPS 1000 Series VoIP Gateway
14.5.
U
55
V
56
W
57
@hinet.net
**01
Region ID
Country
Region ID
Country
Region ID
Australia
02
Korea
24
Philippines
03
Malaysia
26
Canada
06
Singapore
36
China
07
Slovenia
38
Vietnam
10
Spain
40
France
12
Taiwan
43
Germany
13
Thailand
44
Hong Kong
15
British
46
Italy
22
USA
47
Japan
23
58
IPS 1000 Series VoIP Gateway
15.
Contact Information
If you have any question please feel free to ask our technical support, or visit our website
http://www.artdioinc.com.
Taiwan
Add:7th FL.., No.476, Min-Hu RD., Hsin-Chu, Taiwan 300
TEL:+886-3-5295000
FAX:+886-3-5295005
E-Mail:sales@artdioinc.com
Shang Hai
Add:No.8 775 Nong, Hang Dong RD., Shanghai China
TEL:+86-21-64216757 EXT.16
+86-21-64216758 EXT.16
FAX:+86-21-64216758
E-Mail:shsales@artdioinc.com
Los Angles Office
Address:14235 Lomitas Ave. La Puente, CA 91746
TEL:+1-626-336-0369
FAX:+1-626-961-9114
Email:ussales@artdioinc.com
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